Changeset 3457

Timestamp:

Mar 17, 2011 4:34:43 AM (14 years ago)

Author:

bennylp

Message:

Modifications in PJSUA-LIB to support multiple media streams (multiple audio and/or video) and dynamic creation of media transports. This closed #1185 and closed #1201.

1185: Dynamic creation of media transports
============================================
Done:

• media transports are created on demand now

Todo:

• media transport creation is still blocking

1201: Video support in PJSUA-LIB
===================================
Done:

• call now supports N media (N audio and M video)
• number of audio/video streams is configurable per acc
• extra audio stream info in pjsua_call_info to support multiple audio streams

in one call

• video subsys and ffmpeg initialization in PJSUA-LIB
• ability to offer and create video SDP answer
• "dq" for more than 1 audio streams
• introducing pjsua_state and pjsua_get_state()

API change:

• on_stream_created() and on_stream_destroyed() callbacks: changed session to

stream

Todo:

• many others features are disabled, just search for DISABLED_FOR_TICKET_1185

macro (these have also been added to ticket #1193 (Issues & Todos)). Notable
missing features are:

• creation of duplicate SDP m= lines for optional SRTP
• mm.. that's it?

• whole lot of testings

pjsua:
===============

• Added --extra-audio and --video options. Specify these more than once and

each time an extra audio/video streams will be added. :)

Location:

pjproject/branches/projects/2.0-dev

Files:

• build.mak.in (modified) (3 diffs)
• pjmedia/include/pjmedia/transport.h (modified) (1 diff)
• pjsip-apps/src/pjsua/pjsua_app.c (modified) (10 diffs)
• pjsip/build/Makefile (modified) (1 diff)
• pjsip/include/pjsua-lib/pjsua.h (modified) (22 diffs)
• pjsip/include/pjsua-lib/pjsua_internal.h (modified) (6 diffs)
• pjsip/src/pjsua-lib/pjsua_acc.c (modified) (1 diff)
• pjsip/src/pjsua-lib/pjsua_call.c (modified) (27 diffs)
• pjsip/src/pjsua-lib/pjsua_core.c (modified) (13 diffs)
• pjsip/src/pjsua-lib/pjsua_media.c (modified) (43 diffs)

pjproject/branches/projects/2.0-dev/build.mak.in

@@ -59 +59 @@
 @ac_build_mak_vars@
 
+#
+# Video
+# Note: there are duplicated macros in pjmedia/os-auto.mak.in (and that's not
+#       good!
+
+# SDL flags
+SDL_CFLAGS = @ac_sdl_cflags@
+SDL_LDFLAGS = @ac_sdl_ldflags@
+
+# FFMPEG dlags
+FFMPEG_CFLAGS = @ac_ffmpeg_cflags@ 
+FFMPEG_LDFLAGS =  @ac_ffmpeg_ldflags@
+
+# Video4Linux2
+V4L2_CFLAGS = @ac_v4l2_cflags@
+V4L2_LDFLAGS = @ac_v4l2_ldflags@
+
+# QT
+AC_PJMEDIA_VIDEO_HAS_QT = @ac_pjmedia_video_has_qt@
+QT_CFLAGS = @ac_qt_cflags@
+
+# iOS
+IOS_CFLAGS = @ac_ios_cflags@
+
+# PJMEDIA features exclusion
+PJ_VIDEO_CFLAGS += $(SDL_CFLAGS) $(FFMPEG_CFLAGS) $(V4L2_CFLAGS) $(QT_CFLAGS) \
+                   $(IOS_CFLAGS)
+PJ_VIDEO_LDFLAGS += $(SDL_LDFLAGS) $(FFMPEG_LDFLAGS) $(V4L2_LDFLAGS)
+
+
 # CFLAGS, LDFLAGS, and LIBS to be used by applications
 export PJDIR := @ac_pjdir@
@@ -65 +95 @@
 export APP_CFLAGS := -DPJ_AUTOCONF=1\
         @CFLAGS@\
+        $(PJ_VIDEO_CFLAGS) \
         -I$(PJDIR)/pjlib/include\
         -I$(PJDIR)/pjlib-util/include\
@@ -77 +108 @@
         -L$(PJDIR)/pjsip/lib\
         -L$(PJDIR)/third_party/lib\
+        $(PJ_VIDEO_LDFLAGS) \
         @LDFLAGS@
 export APP_LDLIBS := -lpjsua-$(TARGET_NAME)\

pjproject/branches/projects/2.0-dev/pjmedia/include/pjmedia/transport.h

@@ -484 +484 @@
     /** Transport's "virtual" function table. */
     pjmedia_transport_op    *op;
+
+    /** Application/user data */
+    void                    *user_data;
 };
 

pjproject/branches/projects/2.0-dev/pjsip-apps/src/pjsua/pjsua_app.c

@@ -268 +268 @@
 
     puts  ("");
-    puts  ("Media Options:");
+    puts  ("Audio Options:");
     puts  ("  --add-codec=name    Manually add codec (default is to enable all)");
     puts  ("  --dis-codec=name    Disable codec (can be specified multiple times)");
@@ -302 +302 @@
     puts  ("  --no-tones          Disable audible tones");
     puts  ("  --jb-max-size       Specify jitter buffer maximum size, in frames (default=-1)");
+    puts  ("  --extra-audio       Add one more audio stream");
+
+    puts  ("");
+    puts  ("Video Options:");
+    puts  ("  --video             Enable video");
 
     puts  ("");
@@ -541 +546 @@
            OPT_AUTO_UPDATE_NAT,OPT_USE_COMPACT_FORM,OPT_DIS_CODEC,
            OPT_NO_FORCE_LR,
-           OPT_TIMER, OPT_TIMER_SE, OPT_TIMER_MIN_SE
+           OPT_TIMER, OPT_TIMER_SE, OPT_TIMER_MIN_SE,
+           OPT_VIDEO, OPT_EXTRA_AUDIO
     };
     struct pj_getopt_option long_options[] = {
@@ -660 +666 @@
         { "timer-min-se", 1, 0, OPT_TIMER_MIN_SE},
         { "outb-rid",   1, 0, OPT_OUTB_RID},
+        { "video",      0, 0, OPT_VIDEO},
+        { "extra-audio",0, 0, OPT_EXTRA_AUDIO},
         { NULL, 0, 0, 0}
     };
@@ -1416 +1424 @@
             cfg->udp_cfg.qos_params.flags = PJ_QOS_PARAM_HAS_DSCP;
             cfg->udp_cfg.qos_params.dscp_val = 0x18;
+            break;
+        case OPT_VIDEO:
+            ++cur_acc->max_video_cnt;
+            break;
+        case OPT_EXTRA_AUDIO:
+            ++cur_acc->max_audio_cnt;
             break;
         default:
@@ -1661 +1675 @@
     if (acc_cfg->mwi_enabled)
         pj_strcat2(result, "--mwi\n");
+
+    /* Video & extra audio */
+    for (i=0; i<acc_cfg->max_video_cnt; ++i) {
+        pj_strcat2(result, "--video\n");
+    }
+    for (i=1; i<acc_cfg->max_audio_cnt; ++i) {
+        pj_strcat2(result, "--extra-audio\n");
+    }
 }
 
@@ -3892 +3914 @@
                 acc_cfg.cred_info[0].data = pj_str(passwd);
 
+                acc_cfg.rtp_cfg = app_config.rtp_cfg;
+
                 status = pjsua_acc_add(&acc_cfg, PJ_TRUE, NULL);
                 if (status != PJ_SUCCESS) {
@@ -4929 +4953 @@
     /* Add accounts */
     for (i=0; i<app_config.acc_cnt; ++i) {
+        app_config.acc_cfg[i].rtp_cfg = app_config.rtp_cfg;
         status = pjsua_acc_add(&app_config.acc_cfg[i], PJ_TRUE, NULL);
         if (status != PJ_SUCCESS)
@@ -4960 +4985 @@
     if (app_config.ipv6)
         status = create_ipv6_media_transports();
+  #if DISABLED_FOR_TICKET_1185
     else
         status = pjsua_media_transports_create(&app_config.rtp_cfg);
+  #endif
 #endif
     if (status != PJ_SUCCESS)
@@ -5296 +5323 @@
     }
 
+#if DISABLED_FOR_TICKET_1185
     return pjsua_media_transports_attach(tp, i, PJ_TRUE);
-}
-
+#else
+    return PJ_ENOTSUP;
+#endif
+}
+

pjproject/branches/projects/2.0-dev/pjsip/build/Makefile

@@ -77 +77 @@
                         pjsua_acc.o pjsua_call.o pjsua_core.o \
                         pjsua_im.o pjsua_media.o pjsua_pres.o
-export PJSUA_LIB_CFLAGS += $(_CFLAGS)
+export PJSUA_LIB_CFLAGS += $(_CFLAGS) $(PJ_VIDEO_CFLAGS)
 
 

pjproject/branches/projects/2.0-dev/pjsip/include/pjsua-lib/pjsua.h

@@ -251 +251 @@
 #define PJSUA_INVALID_ID            (-1)
 
+/** Disabled features temporarily for media reorganization */
+#define DISABLED_FOR_TICKET_1185        0
+
 /** Call identification */
 typedef int pjsua_call_id;
@@ -282 +285 @@
 #   define PJSUA_ACC_MAX_PROXIES    8
 #endif
-
-#if defined(PJMEDIA_HAS_SRTP) && (PJMEDIA_HAS_SRTP != 0)
 
 /**
@@ -304 +305 @@
 #endif
 
-#endif
-
 /**
  * Controls whether PJSUA-LIB should add ICE media feature tag
@@ -325 +324 @@
 #   define PJSUA_ACQUIRE_CALL_TIMEOUT 2000
 #endif
+
+
+/**
+ * This enumeration represents pjsua state.
+ */
+typedef enum pjsua_state
+{
+    /**
+     * The library has not been initialized.
+     */
+    PJSUA_STATE_NULL,
+
+    /**
+     * After pjsua_create() is called but before pjsua_init() is called.
+     */
+    PJSUA_STATE_CREATED,
+
+    /**
+     * After pjsua_init() is called but before pjsua_start() is called.
+     */
+    PJSUA_STATE_INIT,
+
+    /**
+     * After pjsua_start() is called but before everything is running.
+     */
+    PJSUA_STATE_STARTING,
+
+    /**
+     * After pjsua_start() is called and before pjsua_destroy() is called.
+     */
+    PJSUA_STATE_RUNNING,
+
+    /**
+     * After pjsua_destroy() is called but before the function returns.
+     */
+    PJSUA_STATE_CLOSING
+
+} pjsua_state;
 
 
@@ -484 +521 @@
      *
      * @param call_id       Call identification.
-     * @param sess          Media session for the call.
+     * @param strm          Media stream.
      * @param stream_idx    Stream index in the media session.
      * @param p_port        On input, it specifies the media port of the
@@ -492 +529 @@
      */
     void (*on_stream_created)(pjsua_call_id call_id, 
-                              pjmedia_session *sess,
+                              pjmedia_stream *strm,
                               unsigned stream_idx, 
                               pjmedia_port **p_port);
@@ -501 +538 @@
      *
      * @param call_id       Call identification.
-     * @param sess          Media session for the call.
+     * @param strm          Media stream.
      * @param stream_idx    Stream index in the media session.
      */
     void (*on_stream_destroyed)(pjsua_call_id call_id,
-                                pjmedia_session *sess, 
+                                pjmedia_stream *strm,
                                 unsigned stream_idx);
 
@@ -1155 +1192 @@
     pj_str_t        user_agent;
 
-#if defined(PJMEDIA_HAS_SRTP) && (PJMEDIA_HAS_SRTP != 0)
     /**
      * Specify default value of secure media transport usage. 
@@ -1185 +1221 @@
 
     /**
-     * Specify whether SRTP in PJMEDIA_SRTP_OPTIONAL mode should compose 
+     * Specify whether SRTP in PJMEDIA_SRTP_OPTIONAL mode should compose
      * duplicated media in SDP offer, i.e: unsecured and secured version.
-     * Otherwise, the SDP media will be composed as unsecured media but 
+     * Otherwise, the SDP media will be composed as unsecured media but
      * with SDP "crypto" attribute.
      *
@@ -1193 +1229 @@
      */
     pj_bool_t        srtp_optional_dup_offer;
-#endif
 
     /**
@@ -1351 +1386 @@
  */
 PJ_DECL(pj_status_t) pjsua_destroy(void);
+
+
+/**
+ * Retrieve pjsua state.
+ *
+ * @return      pjsua state.
+ */
+PJ_DECL(pjsua_state) pjsua_get_state(void);
 
 
@@ -2094 +2137 @@
 #endif
 
-
 /**
  * This structure describes account configuration to be specified when
@@ -2385 +2427 @@
     pj_str_t         ka_data;
 
-#if defined(PJMEDIA_HAS_SRTP) && (PJMEDIA_HAS_SRTP != 0)
+    /**
+     * Maximum number of simultaneous active audio streams to be allowed
+     * for calls on this account. Setting this to zero will disable audio
+     * in calls on this account.
+     *
+     * Default: 1
+     */
+    unsigned         max_audio_cnt;
+
+    /**
+     * Maximum number of simultaneous active video streams to be allowed
+     * for calls on this account. Setting this to zero will disable video
+     * in calls on this account.
+     *
+     * Default: 0
+     */
+    unsigned         max_video_cnt;
+
+    /**
+     * Media transport config.
+     */
+    pjsua_transport_config rtp_cfg;
+
     /**
      * Specify whether secure media transport should be used for this account.
@@ -2409 +2473 @@
 
     /**
-     * Specify whether SRTP in PJMEDIA_SRTP_OPTIONAL mode should compose 
+     * Specify whether SRTP in PJMEDIA_SRTP_OPTIONAL mode should compose
      * duplicated media in SDP offer, i.e: unsecured and secured version.
-     * Otherwise, the SDP media will be composed as unsecured media but 
+     * Otherwise, the SDP media will be composed as unsecured media but
      * with SDP "crypto" attribute.
      *
@@ -2417 +2481 @@
      */
     pj_bool_t        srtp_optional_dup_offer;
-#endif
 
     /**
@@ -2921 +2984 @@
 typedef enum pjsua_call_media_status
 {
-    /** Call currently has no media */
+    /**
+     * Call currently has no media, or the media is not used.
+     */
     PJSUA_CALL_MEDIA_NONE,
 
-    /** The media is active */
+    /**
+     * The media is active
+     */
     PJSUA_CALL_MEDIA_ACTIVE,
 
-    /** The media is currently put on hold by local endpoint */
+    /**
+     * The media is currently put on hold by local endpoint
+     */
     PJSUA_CALL_MEDIA_LOCAL_HOLD,
 
-    /** The media is currently put on hold by remote endpoint */
+    /**
+     * The media is currently put on hold by remote endpoint
+     */
     PJSUA_CALL_MEDIA_REMOTE_HOLD,
 
-    /** The media has reported error (e.g. ICE negotiation) */
+    /**
+     * The media has reported error (e.g. ICE negotiation)
+     */
     PJSUA_CALL_MEDIA_ERROR
 
@@ -2983 +3056 @@
     pjsua_call_media_status media_status;
 
-    /** Media direction */
+    /** Number of active audio streams in this call */
+    unsigned            audio_cnt;
+
+    /** Media direction of the first audio stream. */
     pjmedia_dir         media_dir;
 
-    /** The conference port number for the call */
+    /** The conference port number for the first audio stream. */
     pjsua_conf_port_id  conf_slot;
+
+    /** Array of audio media stream information */
+    struct
+    {
+        /** Media index in SDP. */
+        unsigned                index;
+
+        /** Call media status. */
+        pjsua_call_media_status media_status;
+
+        /** Media direction. */
+        pjmedia_dir             media_dir;
+
+        /** The conference port number for the call. */
+        pjsua_conf_port_id      conf_slot;
+
+    } audio[4];
 
     /** Up-to-date call connected duration (zero when call is not 
@@ -3083 +3176 @@
 
 
+#if DISABLED_FOR_TICKET_1185
 /**
  * Retrieve the media session associated with this call. Note that the media
@@ -3096 +3190 @@
 PJ_DECL(pjmedia_session*) pjsua_call_get_media_session(pjsua_call_id call_id);
 
-
 /**
  * Retrieve the media transport instance that is used for this call. 
@@ -3107 +3200 @@
  */
 PJ_DECL(pjmedia_transport*) pjsua_call_get_media_transport(pjsua_call_id cid);
-
+#endif /* DISABLED_FOR_TICKET_1185 */
 
 /**
@@ -4883 +4976 @@
                                             const pjmedia_codec_param *param);
 
-
-
-
+#if DISABLED_FOR_TICKET_1185
 /**
  * Create UDP media transports for all the calls. This function creates
@@ -4917 +5008 @@
                                unsigned count,
                                pj_bool_t auto_delete);
+#endif
 
 

pjproject/branches/projects/2.0-dev/pjsip/include/pjsua-lib/pjsua_internal.h

@@ -44 +44 @@
 } pjsua_med_tp_st;
 
+/** Forward decl of pjsua call */
+typedef struct pjsua_call pjsua_call;
+
+/**
+ * Call's media stream.
+ */
+typedef struct pjsua_call_media
+{
+    pjsua_call          *call;      /**< Parent call.                       */
+    pjmedia_type         type;      /**< Media type.                        */
+    unsigned             idx;       /**< This media index in parent call.   */
+    pjsua_call_media_status state;  /**< Media state.                       */
+    pjmedia_dir          dir;       /**< Media direction.                   */
+
+    /** The stream */
+    union {
+        /** Audio stream */
+        struct {
+            pjmedia_stream *stream;    /**< The media session.              */
+            int             conf_slot; /**< Slot # in conference bridge.    */
+        } a;
+
+        /** Video stream */
+        struct {
+        } v;
+
+    } strm;
+
+    pj_uint32_t          ssrc;      /**< RTP SSRC                           */
+    pj_uint32_t          rtp_tx_ts; /**< Initial RTP timestamp for sender.  */
+    pj_uint16_t          rtp_tx_seq;/**< Initial RTP sequence for sender.   */
+    pj_uint8_t           rtp_tx_seq_ts_set;
+                                    /**< Bitmask flags if initial RTP sequence
+                                         and/or timestamp for sender are set.
+                                         bit 0/LSB : sequence flag
+                                         bit 1     : timestamp flag         */
+
+    pjmedia_transport   *tp;        /**< Current media transport (can be 0) */
+    pj_status_t          tp_ready;  /**< Media transport status.            */
+    pjmedia_transport   *tp_orig;   /**< Original media transport           */
+    pj_bool_t            tp_auto_del; /**< May delete media transport   */
+    pjsua_med_tp_st      tp_st;     /**< Media transport state              */
+    pj_sockaddr          rtp_addr;  /**< Current RTP source address
+                                            (used to update ICE default
+                                            address)                        */
+    pjmedia_srtp_use     rem_srtp_use; /**< Remote's SRTP usage policy.     */
+} pjsua_call_media;
+
+/**
+ * Maximum number of SDP "m=" lines to be supported.
+ */
+#define PJSUA_MAX_CALL_MEDIA            PJMEDIA_MAX_SDP_MEDIA
+
 /** 
  * Structure to be attached to invite dialog. 
@@ -49 +102 @@
  * by accessing dlg->mod_data[pjsua.mod.id].
  */
-typedef struct pjsua_call
+struct pjsua_call
 {
     unsigned             index;     /**< Index in pjsua array.              */
@@ -64 +117 @@
     pjsua_call_hold_type call_hold_type; /**< How to do call hold.          */
     pj_bool_t            local_hold;/**< Flag for call-hold by local.       */
-    pjsua_call_media_status media_st;/**< Media state.                      */
-    pjmedia_dir          media_dir; /**< Media direction.                   */
-    pjmedia_session     *session;   /**< The media session.                 */
-    int                  audio_idx; /**< Index of m=audio in SDP.           */
-    pj_uint32_t          ssrc;      /**< RTP SSRC                           */
-    pj_uint32_t          rtp_tx_ts; /**< Initial RTP timestamp for sender.  */
-    pj_uint16_t          rtp_tx_seq;/**< Initial RTP sequence for sender.   */
-    pj_uint8_t           rtp_tx_seq_ts_set;
-                                    /**< Bitmask flags if initial RTP sequence
-                                         and/or timestamp for sender are set.
-                                         bit 0/LSB : sequence flag 
-                                         bit 1     : timestamp flag         */
-    int                  conf_slot; /**< Slot # in conference bridge.       */
+
+    unsigned             med_cnt;   /**< Number of media in SDP.            */
+    pjsua_call_media     media[PJSUA_MAX_CALL_MEDIA]; /**< Array of media   */
+    unsigned             audio_idx; /**< Selected audio media.              */
+
     pjsip_evsub         *xfer_sub;  /**< Xfer server subscription, if this
                                          call was triggered by xfer.        */
-    pjmedia_transport   *med_tp;    /**< Current media transport.           */
-    pj_status_t          med_tp_ready;/**< Media transport status.          */
-    pjmedia_transport   *med_orig;  /**< Original media transport           */
-    pj_bool_t            med_tp_auto_del; /**< May delete media transport   */
-    pjsua_med_tp_st      med_tp_st; /**< Media transport state              */
-    pj_sockaddr          med_rtp_addr; /**< Current RTP source address
-                                            (used to update ICE default
-                                            address)                        */
     pj_stun_nat_type     rem_nat_type; /**< NAT type of remote endpoint.    */
-    pjmedia_srtp_use     rem_srtp_use; /**< Remote's SRTP usage policy.     */
 
     char    last_text_buf_[128];    /**< Buffer for last_text.              */
@@ -100 +136 @@
                                           contains multiple codecs.         */
 
-} pjsua_call;
+};
 
 
@@ -271 +307 @@
     pj_pool_t           *pool;      /**< pjsua's private pool.          */
     pj_mutex_t          *mutex;     /**< Mutex protection for this data */
+    pjsua_state          state;     /**< Library state.                 */
 
     /* Logging: */
@@ -402 +439 @@
 #define PJSUA_UNLOCK()
 #endif
+
+/* Core */
+void pjsua_set_state(pjsua_state new_state);
 
 /******

pjproject/branches/projects/2.0-dev/pjsip/src/pjsua-lib/pjsua_acc.c

@@ -119 +119 @@
     pjsip_auth_clt_pref_dup(pool, &dst->auth_pref, &src->auth_pref);
 
-    dst->ka_interval = src->ka_interval;
+    pjsua_transport_config_dup(pool, &dst->rtp_cfg, &src->rtp_cfg);
+
     pj_strdup(pool, &dst->ka_data, &src->ka_data);
 }

pjproject/branches/projects/2.0-dev/pjsip/src/pjsua-lib/pjsua_call.c

@@ -102 +102 @@
 {
     pjsua_call *call = &pjsua_var.calls[id];
-
+    unsigned i;
+
+    pj_bzero(call, sizeof(*call));
     call->index = id;
-    call->inv = NULL;
-    call->user_data = NULL;
-    call->session = NULL;
-    call->audio_idx = -1;
-    call->ssrc = pj_rand();
-    call->rtp_tx_seq = 0;
-    call->rtp_tx_ts = 0;
-    call->rtp_tx_seq_ts_set = 0;
-    call->xfer_sub = NULL;
-    call->last_code = (pjsip_status_code) 0;
-    call->conf_slot = PJSUA_INVALID_ID;
     call->last_text.ptr = call->last_text_buf_;
-    call->last_text.slen = 0;
-    call->conn_time.sec = 0;
-    call->conn_time.msec = 0;
-    call->res_time.sec = 0;
-    call->res_time.msec = 0;
-    call->rem_nat_type = PJ_STUN_NAT_TYPE_UNKNOWN;
-    call->rem_srtp_use = PJMEDIA_SRTP_DISABLED;
-    call->local_hold = PJ_FALSE;
-    pj_bzero(&call->lock_codec, sizeof(call->lock_codec));
+    for (i=0; i<PJ_ARRAY_SIZE(call->media); ++i) {
+        pjsua_call_media *call_med = &call->media[i];
+        call_med->ssrc = pj_rand();
+        call_med->strm.a.conf_slot = PJSUA_INVALID_ID;
+        call_med->call = call;
+        call_med->idx = i;
+        call_med->tp_auto_del = PJ_TRUE;
+    }
 }
 
@@ -824 +814 @@
     status = pjsua_media_channel_init(call->index, PJSIP_ROLE_UAS, 
                                       call->secure_level, 
-                                      rdata->tp_info.pool, offer,
+                                      rdata->tp_info.pool,
+                                      offer,
                                       &sip_err_code);
     if (status != PJ_SUCCESS) {
@@ -1119 +1110 @@
 PJ_DEF(pj_bool_t) pjsua_call_has_media(pjsua_call_id call_id)
 {
+    pjsua_call *call = &pjsua_var.calls[call_id];
     PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls, 
                      PJ_EINVAL);
-    return pjsua_var.calls[call_id].session != NULL;
-}
-
-
+    return call->audio_idx >= 0 && call->media[call->audio_idx].strm.a.stream;
+}
+
+
+#if DISABLED_FOR_TICKET_1185
 /*
  * Retrieve the media session associated with this call.
@@ -1143 +1136 @@
     PJ_ASSERT_RETURN(cid>=0 && cid<(int)pjsua_var.ua_cfg.max_calls, 
                      NULL);
-    return pjsua_var.calls[cid].med_tp;
-}
-
+    return pjsua_var.calls[cid].tp;
+}
+#endif /* Removed in 2.0 */
 
 /* Acquire lock to the specified call_id */
@@ -1239 +1232 @@
         return PJSUA_INVALID_ID;
 
-    port_id = call->conf_slot;
+    port_id = call->media[call->audio_idx].strm.a.conf_slot;
 
     pjsip_dlg_dec_lock(dlg);
@@ -1256 +1249 @@
     pjsua_call *call;
     pjsip_dialog *dlg;
+    unsigned mi;
     pj_status_t status;
 
@@ -1330 +1324 @@
     }
     
-    /* media status and dir */
-    info->media_status = call->media_st;
-    info->media_dir = call->media_dir;
-
+    /* Build array of media status and dir */
+    info->audio_cnt = 0;
+    for (mi=0; mi < call->med_cnt &&
+               info->audio_cnt < PJ_ARRAY_SIZE(info->audio); ++mi)
+    {
+        pjsua_call_media *call_med = &call->media[mi];
+        if (call_med->type != PJMEDIA_TYPE_AUDIO)
+            continue;
+        info->audio[info->audio_cnt].index = mi;
+        info->audio[info->audio_cnt].media_status = call_med->state;
+        info->audio[info->audio_cnt].media_dir = call_med->dir;
+        info->audio[info->audio_cnt].conf_slot = call_med->strm.a.conf_slot;
+        ++info->audio_cnt;
+    }
+
+    if (info->audio_cnt) {
+        info->media_status = info->audio[0].media_status;
+        info->media_dir = info->audio[0].media_dir;
+    }
 
     /* conference slot number */
-    info->conf_slot = call->conf_slot;
+    info->conf_slot = call->media[call->audio_idx].strm.a.conf_slot;
 
     /* calculate duration */
@@ -1960 +1969 @@
         return status;
 
-    if (!call->session) {
+    if (!pjsua_call_has_media(call_id)) {
         PJ_LOG(3,(THIS_FILE, "Media is not established yet!"));
         pjsip_dlg_dec_lock(dlg);
@@ -1966 +1975 @@
     }
 
-    status = pjmedia_session_dial_dtmf( call->session, 0, digits);
+    status = pjmedia_stream_dial_dtmf(
+                call->media[call->audio_idx].strm.a.stream, digits);
 
     pjsip_dlg_dec_lock(dlg);
@@ -2191 +2201 @@
     char *p = buf, *end = buf+maxlen;
     int len;
-    pjmedia_session_info info;
-    pjmedia_session *session = call->session;
-    pjmedia_transport_info tp_info;
-
-    pjmedia_transport_info_init(&tp_info);
-
-    pjmedia_transport_get_info(call->med_tp, &tp_info);
-    pjmedia_session_get_info(session, &info);
-
-    for (i=0; i<info.stream_cnt; ++i) {
+
+    for (i=0; i<call->med_cnt; ++i) {
+        pjsua_call_media *call_med = &call->media[i];
+        pjmedia_stream_info info;
+        pjmedia_stream *stream = call_med->strm.a.stream;
+        pjmedia_transport_info tp_info;
         pjmedia_rtcp_stat stat;
         char rem_addr_buf[80];
@@ -2209 +2215 @@
         pj_time_val media_duration, now;
 
-        pjmedia_session_get_stream_stat(session, i, &stat);
+        /* Check if the stream is deactivated */
+        if (call_med->tp == NULL || stream == NULL) {
+            const char *media_type_str;
+
+            switch (call_med->type) {
+            case PJMEDIA_TYPE_AUDIO:
+                media_type_str = "audio";
+                break;
+            case PJMEDIA_TYPE_VIDEO:
+                media_type_str = "video";
+                break;
+            case PJMEDIA_TYPE_APPLICATION:
+                media_type_str = "application";
+                break;
+            default:
+                media_type_str = "unknown";
+                break;
+            }
+            len = pj_ansi_snprintf(p, end-p,
+                      "%s  #%d m=%s deactivated\n",
+                      indent, i, media_type_str);
+            if (len < 1 || len > end-p) {
+                *p = '\0';
+                return;
+            }
+
+            p += len;
+            continue;
+        }
+
+        pjmedia_transport_info_init(&tp_info);
+        pjmedia_transport_get_info(call_med->tp, &tp_info);
+
+        pjmedia_stream_get_info(stream, &info);
+        pjmedia_stream_get_stat(stream, &stat);
+
         // rem_addr will contain actual address of RTP originator, instead of
         // remote RTP address specified by stream which is fetched from the SDP.
@@ -2223 +2264 @@
         }
 
-        if (call->media_dir == PJMEDIA_DIR_NONE) {
+        if (call_med->dir == PJMEDIA_DIR_NONE) {
             /* To handle when the stream that is currently being paused
              * (http://trac.pjsip.org/repos/ticket/1079)
              */
             dir = "inactive";
-        } else if (info.stream_info[i].dir == PJMEDIA_DIR_ENCODING)
+        } else if (info.dir == PJMEDIA_DIR_ENCODING)
             dir = "sendonly";
-        else if (info.stream_info[i].dir == PJMEDIA_DIR_DECODING)
+        else if (info.dir == PJMEDIA_DIR_DECODING)
             dir = "recvonly";
-        else if (info.stream_info[i].dir == PJMEDIA_DIR_ENCODING_DECODING)
+        else if (info.dir == PJMEDIA_DIR_ENCODING_DECODING)
             dir = "sendrecv";
         else
@@ -2238 +2279 @@
 
         
-        len = pj_ansi_snprintf(buf, end-p, 
+        len = pj_ansi_snprintf(p, end-p,
                   "%s  #%d %.*s @%dKHz, %s, peer=%s",
                   indent, i,
-                  (int)info.stream_info[i].fmt.encoding_name.slen,
-                  info.stream_info[i].fmt.encoding_name.ptr,
-                  info.stream_info[i].fmt.clock_rate / 1000,
+                  (int)info.fmt.encoding_name.slen,
+                  info.fmt.encoding_name.ptr,
+                  info.fmt.clock_rate / 1000,
                   dir,
                   rem_addr);
@@ -2294 +2335 @@
 #endif
                "%s",
-               indent, info.stream_info[i].fmt.pt,
+               indent, info.fmt.pt,
                last_update,
                indent,
@@ -2371 +2412 @@
                "%s        jitter     : %7.3f %7.3f %7.3f %7.3f %7.3f%s",
                indent,
-               info.stream_info[i].tx_pt,
-               info.stream_info[i].param->info.frm_ptime *
-                info.stream_info[i].param->setting.frm_per_pkt,
+               info.tx_pt,
+               info.param->info.frm_ptime * info.param->setting.frm_per_pkt,
                last_update,
 
@@ -2416 +2456 @@
 
         len = pj_ansi_snprintf(p, end-p,
-               "%s    RTT msec       : %7.3f %7.3f %7.3f %7.3f %7.3f", 
+               "%s     RTT msec      : %7.3f %7.3f %7.3f %7.3f %7.3f",
                indent,
                stat.rtt.min / 1000.0,
@@ -2471 +2511 @@
             }
 
-            clock_rate = info.stream_info[i].fmt.clock_rate;
+            clock_rate = info.fmt.clock_rate;
 
             len = pj_ansi_snprintf(p, end-p, "\n%s  Extended reports:", indent);
@@ -2944 +2984 @@
 
     /* Get and ICE SRTP status */
+#if DISABLED_FOR_TICKET_1185
     pjmedia_transport_info_init(&tp_info);
-    pjmedia_transport_get_info(call->med_tp, &tp_info);
+    pjmedia_transport_get_info(call->tp, &tp_info);
     if (tp_info.specific_info_cnt > 0) {
         unsigned i;
@@ -2984 +3025 @@
         }
     }
+#endif  /* DISABLED_FOR_TICKET_1185 */
 
     /* Dump session statistics */
-    if (with_media && call->session)
+    if (with_media && pjsua_call_has_media(call_id))
         dump_media_session(indent, p, end-p, call);
 
@@ -3064 +3106 @@
     const pjmedia_sdp_media *ref_m;
     pjmedia_sdp_media *m;
+    pjsua_call_media *call_med = &call->media[call->audio_idx];
     unsigned i, codec_cnt = 0;
     pj_bool_t rem_can_update;
@@ -3098 +3141 @@
 
     /* Verify if media is deactivated */
-    if (call->media_st == PJSUA_CALL_MEDIA_NONE ||
-        call->media_st == PJSUA_CALL_MEDIA_ERROR ||
-        call->media_dir == PJMEDIA_DIR_NONE)
+    if (call_med->state == PJSUA_CALL_MEDIA_NONE ||
+        call_med->state == PJSUA_CALL_MEDIA_ERROR ||
+        call_med->dir == PJMEDIA_DIR_NONE)
     {
         return PJ_EINVALIDOP;
@@ -3190 +3233 @@
     const pjmedia_sdp_media *rem_m, *loc_m;
     unsigned codec_cnt=0, i;
+    pjsua_call_media *call_med = &call->media[call->audio_idx];
     pj_time_val delay = {0, 0};
     const pj_str_t st_update = {"UPDATE", 6};
@@ -3207 +3251 @@
 
     /* Skip this if the media is inactive or error */
-    if (call->media_st == PJSUA_CALL_MEDIA_NONE ||
-        call->media_st == PJSUA_CALL_MEDIA_ERROR ||
-        call->media_dir == PJMEDIA_DIR_NONE)
+    if (call_med->state == PJSUA_CALL_MEDIA_NONE ||
+        call_med->state == PJSUA_CALL_MEDIA_ERROR ||
+        call_med->dir == PJMEDIA_DIR_NONE)
     {
         return PJ_SUCCESS;
@@ -3525 +3569 @@
 
     /* Add SDP in 488 status */
-    if (call && call->med_tp && tdata->msg->type==PJSIP_RESPONSE_MSG && 
+#if DISABLED_FOR_TICKET_1185
+    if (call && call->tp && tdata->msg->type==PJSIP_RESPONSE_MSG &&
         code==PJSIP_SC_NOT_ACCEPTABLE_HERE) 
     {
@@ -3540 +3585 @@
         }
     }
+#endif
 
     pjsip_inv_send_msg(inv, tdata);
@@ -3655 +3701 @@
      */
     /* http://trac.pjsip.org/repos/ticket/880 
-       if (call->media_dir != PJMEDIA_DIR_ENCODING) {
+       if (call->dir != PJMEDIA_DIR_ENCODING) {
      */
     /* https://trac.pjsip.org/repos/ticket/1142:
@@ -3695 +3741 @@
         pjmedia_sdp_media_remove_all_attr(m, "inactive");
 
-        if (call->media_dir & PJMEDIA_DIR_ENCODING) {
+        if (call->media[call->audio_idx].dir & PJMEDIA_DIR_ENCODING) {
             /* Add sendonly attribute */
             attr = pjmedia_sdp_attr_create(pool, "sendonly", NULL);

pjproject/branches/projects/2.0-dev/pjsip/src/pjsua-lib/pjsua_core.c

@@ -104 +104 @@
     cfg->force_lr = PJ_TRUE;
     cfg->enable_unsolicited_mwi = PJ_TRUE;
-#if defined(PJMEDIA_HAS_SRTP) && (PJMEDIA_HAS_SRTP != 0)
     cfg->use_srtp = PJSUA_DEFAULT_USE_SRTP;
     cfg->srtp_secure_signaling = PJSUA_DEFAULT_SRTP_SECURE_SIGNALING;
-#endif
     cfg->hangup_forked_call = PJ_TRUE;
 
@@ -177 +175 @@
     cfg->ka_interval = 15;
     cfg->ka_data = pj_str("\r\n");
-#if defined(PJMEDIA_HAS_SRTP) && (PJMEDIA_HAS_SRTP != 0)
+    cfg->max_audio_cnt = 1;
+    pjsua_transport_config_default(&cfg->rtp_cfg);
     cfg->use_srtp = pjsua_var.ua_cfg.use_srtp;
     cfg->srtp_secure_signaling = pjsua_var.ua_cfg.srtp_secure_signaling;
     cfg->srtp_optional_dup_offer = pjsua_var.ua_cfg.srtp_optional_dup_offer;
-#endif
     cfg->reg_retry_interval = PJSUA_REG_RETRY_INTERVAL;
     cfg->contact_rewrite_method = PJSUA_CONTACT_REWRITE_METHOD;
@@ -224 +222 @@
 }
 
-
 /*****************************************************************************
  * This is a very simple PJSIP module, whose sole purpose is to display
@@ -379 +376 @@
 
     /* Get media socket info, make sure transport is ready */
+#if DISABLED_FOR_TICKET_1185
     if (pjsua_var.calls[0].med_tp) {
         pjmedia_transport_info_init(&tpinfo);
@@ -390 +388 @@
         }
     }
-
-    /* Send response statelessly */
+#endif
+
+    /* Send response */
     pjsip_get_response_addr(tdata->pool, rdata, &res_addr);
     status = pjsip_endpt_send_response(pjsua_var.endpt, &res_addr, tdata, NULL, NULL);
@@ -660 +659 @@
     PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);
 
+    pjsua_set_state(PJSUA_STATE_CREATED);
 
     return PJ_SUCCESS;
@@ -919 +919 @@
                          pj_get_version(), PJ_OS_NAME));
 
+    pjsua_set_state(PJSUA_STATE_INIT);
+
     return PJ_SUCCESS;
 
@@ -1276 +1278 @@
     int i;  /* Must be signed */
 
+    if (pjsua_var.state > PJSUA_STATE_NULL &&
+        pjsua_var.state < PJSUA_STATE_CLOSING)
+    {
+        pjsua_set_state(PJSUA_STATE_CLOSING);
+    }
+
     /* Signal threads to quit: */
     pjsua_var.thread_quit_flag = 1;
@@ -1453 +1461 @@
     pj_bzero(&pjsua_var, sizeof(pjsua_var));
 
+    pjsua_set_state(PJSUA_STATE_NULL);
+
     /* Done. */
     return PJ_SUCCESS;
 }
 
+void pjsua_set_state(pjsua_state new_state)
+{
+    const char *state_name[] = {
+        "NULL",
+        "CREATED",
+        "INIT",
+        "STARTING",
+        "RUNNING",
+        "CLOSING"
+    };
+    pjsua_state old_state = pjsua_var.state;
+
+    pjsua_var.state = new_state;
+    PJ_LOG(4,(THIS_FILE, "PJSUA state changed: %s --> %s",
+              state_name[old_state], state_name[new_state]));
+}
+
+/* Get state */
+PJ_DEF(pjsua_state) pjsua_get_state(void)
+{
+    return pjsua_var.state;
+}
 
 /**
@@ -1469 +1501 @@
     pj_status_t status;
 
+    pjsua_set_state(PJSUA_STATE_STARTING);
+
     status = pjsua_call_subsys_start();
     if (status != PJ_SUCCESS)
@@ -1480 +1514 @@
     if (status != PJ_SUCCESS)
         return status;
+
+    pjsua_set_state(PJSUA_STATE_RUNNING);
 
     return PJ_SUCCESS;
@@ -2564 +2600 @@
 
     PJ_LOG(3,(THIS_FILE, "Dumping media transports:"));
+#if DISABLED_FOR_TICKET_1185
     for (i=0; i<pjsua_var.ua_cfg.max_calls; ++i) {
         pjsua_call *call = &pjsua_var.calls[i];
@@ -2580 +2617 @@
                                     sizeof(addr_buf), 3)));
     }
+#endif
 
     pjsip_tsx_layer_dump(detail);

pjproject/branches/projects/2.0-dev/pjsip/src/pjsua-lib/pjsua_media.c

@@ -275 +275 @@
 #endif  /* PJMEDIA_HAS_L16_CODEC */
 
+#if PJMEDIA_HAS_VIDEO
+    status = pjmedia_video_format_mgr_create(pjsua_var.pool, 64, 0, NULL);
+    if (status != PJ_SUCCESS) {
+        pjsua_perror(THIS_FILE, "Error creating PJMEDIA video format manager",
+                     status);
+        return status;
+    }
+
+    status = pjmedia_converter_mgr_create(pjsua_var.pool, NULL);
+    if (status != PJ_SUCCESS) {
+        pjsua_perror(THIS_FILE, "Error creating PJMEDIA converter manager",
+                     status);
+        return status;
+    }
+
+    status = pjmedia_vid_codec_mgr_create(pjsua_var.pool, NULL);
+    if (status != PJ_SUCCESS) {
+        pjsua_perror(THIS_FILE, "Error creating PJMEDIA video codec manager",
+                     status);
+        return status;
+    }
+
+    status = pjmedia_vid_subsys_init(&pjsua_var.cp.factory);
+    if (status != PJ_SUCCESS) {
+        pjsua_perror(THIS_FILE, "Error creating PJMEDIA video subsystem",
+                     status);
+        return status;
+    }
+#endif
+
+#if PJMEDIA_HAS_VIDEO && PJMEDIA_HAS_FFMPEG_CODEC
+    /* Init ffmpeg video codecs */
+    status = pjmedia_codec_ffmpeg_init(NULL, &pjsua_var.cp.factory);
+    if (status != PJ_SUCCESS) {
+        pjsua_perror(THIS_FILE, "Error initializing ffmpeg library",
+                     status);
+        return status;
+    }
+#endif
 
     /* Save additional conference bridge parameters for future
@@ -339 +378 @@
 
 
-/* 
- * Create RTP and RTCP socket pair, and possibly resolve their public
- * address via STUN.
- */
-static pj_status_t create_rtp_rtcp_sock(const pjsua_transport_config *cfg,
-                                        pjmedia_sock_info *skinfo)
-{
-    enum { 
-        RTP_RETRY = 100
-    };
-    int i;
-    pj_sockaddr_in bound_addr;
-    pj_sockaddr_in mapped_addr[2];
-    pj_status_t status = PJ_SUCCESS;
-    char addr_buf[PJ_INET6_ADDRSTRLEN+2];
-    pj_sock_t sock[2];
-
-    /* Make sure STUN server resolution has completed */
-    status = resolve_stun_server(PJ_TRUE);
-    if (status != PJ_SUCCESS) {
-        pjsua_perror(THIS_FILE, "Error resolving STUN server", status);
-        return status;
-    }
-
-    if (next_rtp_port == 0)
-        next_rtp_port = (pj_uint16_t)cfg->port;
-
-    for (i=0; i<2; ++i)
-        sock[i] = PJ_INVALID_SOCKET;
-
-    bound_addr.sin_addr.s_addr = PJ_INADDR_ANY;
-    if (cfg->bound_addr.slen) {
-        status = pj_sockaddr_in_set_str_addr(&bound_addr, &cfg->bound_addr);
-        if (status != PJ_SUCCESS) {
-            pjsua_perror(THIS_FILE, "Unable to resolve transport bind address",
-                         status);
-            return status;
-        }
-    }
-
-    /* Loop retry to bind RTP and RTCP sockets. */
-    for (i=0; i<RTP_RETRY; ++i, next_rtp_port += 2) {
-
-        /* Create RTP socket. */
-        status = pj_sock_socket(pj_AF_INET(), pj_SOCK_DGRAM(), 0, &sock[0]);
-        if (status != PJ_SUCCESS) {
-            pjsua_perror(THIS_FILE, "socket() error", status);
-            return status;
-        }
-
-        /* Apply QoS to RTP socket, if specified */
-        status = pj_sock_apply_qos2(sock[0], cfg->qos_type, 
-                                    &cfg->qos_params, 
-                                    2, THIS_FILE, "RTP socket");
-
-        /* Bind RTP socket */
-        status=pj_sock_bind_in(sock[0], pj_ntohl(bound_addr.sin_addr.s_addr), 
-                               next_rtp_port);
-        if (status != PJ_SUCCESS) {
-            pj_sock_close(sock[0]); 
-            sock[0] = PJ_INVALID_SOCKET;
-            continue;
-        }
-
-        /* Create RTCP socket. */
-        status = pj_sock_socket(pj_AF_INET(), pj_SOCK_DGRAM(), 0, &sock[1]);
-        if (status != PJ_SUCCESS) {
-            pjsua_perror(THIS_FILE, "socket() error", status);
-            pj_sock_close(sock[0]);
-            return status;
-        }
-
-        /* Apply QoS to RTCP socket, if specified */
-        status = pj_sock_apply_qos2(sock[1], cfg->qos_type, 
-                                    &cfg->qos_params, 
-                                    2, THIS_FILE, "RTCP socket");
-
-        /* Bind RTCP socket */
-        status=pj_sock_bind_in(sock[1], pj_ntohl(bound_addr.sin_addr.s_addr), 
-                               (pj_uint16_t)(next_rtp_port+1));
-        if (status != PJ_SUCCESS) {
-            pj_sock_close(sock[0]); 
-            sock[0] = PJ_INVALID_SOCKET;
-
-            pj_sock_close(sock[1]); 
-            sock[1] = PJ_INVALID_SOCKET;
-            continue;
-        }
-
-        /*
-         * If we're configured to use STUN, then find out the mapped address,
-         * and make sure that the mapped RTCP port is adjacent with the RTP.
-         */
-        if (pjsua_var.stun_srv.addr.sa_family != 0) {
-            char ip_addr[32];
-            pj_str_t stun_srv;
-
-            pj_ansi_strcpy(ip_addr, 
-                           pj_inet_ntoa(pjsua_var.stun_srv.ipv4.sin_addr));
-            stun_srv = pj_str(ip_addr);
-
-            status=pjstun_get_mapped_addr(&pjsua_var.cp.factory, 2, sock,
-                                           &stun_srv, pj_ntohs(pjsua_var.stun_srv.ipv4.sin_port),
-                                           &stun_srv, pj_ntohs(pjsua_var.stun_srv.ipv4.sin_port),
-                                           mapped_addr);
-            if (status != PJ_SUCCESS) {
-                pjsua_perror(THIS_FILE, "STUN resolve error", status);
-                goto on_error;
-            }
-
-#if PJSUA_REQUIRE_CONSECUTIVE_RTCP_PORT
-            if (pj_ntohs(mapped_addr[1].sin_port) == 
-                pj_ntohs(mapped_addr[0].sin_port)+1)
-            {
-                /* Success! */
-                break;
-            }
-
-            pj_sock_close(sock[0]); 
-            sock[0] = PJ_INVALID_SOCKET;
-
-            pj_sock_close(sock[1]); 
-            sock[1] = PJ_INVALID_SOCKET;
-#else
-            if (pj_ntohs(mapped_addr[1].sin_port) != 
-                pj_ntohs(mapped_addr[0].sin_port)+1)
-            {
-                PJ_LOG(4,(THIS_FILE, 
-                          "Note: STUN mapped RTCP port %d is not adjacent"
-                          " to RTP port %d",
-                          pj_ntohs(mapped_addr[1].sin_port),
-                          pj_ntohs(mapped_addr[0].sin_port)));
-            }
-            /* Success! */
-            break;
-#endif
-
-        } else if (cfg->public_addr.slen) {
-
-            status = pj_sockaddr_in_init(&mapped_addr[0], &cfg->public_addr,
-                                         (pj_uint16_t)next_rtp_port);
-            if (status != PJ_SUCCESS)
-                goto on_error;
-
-            status = pj_sockaddr_in_init(&mapped_addr[1], &cfg->public_addr,
-                                         (pj_uint16_t)(next_rtp_port+1));
-            if (status != PJ_SUCCESS)
-                goto on_error;
-
-            break;
-
-        } else {
-
-            if (bound_addr.sin_addr.s_addr == 0) {
-                pj_sockaddr addr;
-
-                /* Get local IP address. */
-                status = pj_gethostip(pj_AF_INET(), &addr);
-                if (status != PJ_SUCCESS)
-                    goto on_error;
-
-                bound_addr.sin_addr.s_addr = addr.ipv4.sin_addr.s_addr;
-            }
-
-            for (i=0; i<2; ++i) {
-                pj_sockaddr_in_init(&mapped_addr[i], NULL, 0);
-                mapped_addr[i].sin_addr.s_addr = bound_addr.sin_addr.s_addr;
-            }
-
-            mapped_addr[0].sin_port=pj_htons((pj_uint16_t)next_rtp_port);
-            mapped_addr[1].sin_port=pj_htons((pj_uint16_t)(next_rtp_port+1));
-            break;
-        }
-    }
-
-    if (sock[0] == PJ_INVALID_SOCKET) {
-        PJ_LOG(1,(THIS_FILE, 
-                  "Unable to find appropriate RTP/RTCP ports combination"));
-        goto on_error;
-    }
-
-
-    skinfo->rtp_sock = sock[0];
-    pj_memcpy(&skinfo->rtp_addr_name, 
-              &mapped_addr[0], sizeof(pj_sockaddr_in));
-
-    skinfo->rtcp_sock = sock[1];
-    pj_memcpy(&skinfo->rtcp_addr_name, 
-              &mapped_addr[1], sizeof(pj_sockaddr_in));
-
-    PJ_LOG(4,(THIS_FILE, "RTP socket reachable at %s",
-              pj_sockaddr_print(&skinfo->rtp_addr_name, addr_buf,
-                                sizeof(addr_buf), 3)));
-    PJ_LOG(4,(THIS_FILE, "RTCP socket reachable at %s",
-              pj_sockaddr_print(&skinfo->rtcp_addr_name, addr_buf,
-                                sizeof(addr_buf), 3)));
-
-    next_rtp_port += 2;
-    return PJ_SUCCESS;
-
-on_error:
-    for (i=0; i<2; ++i) {
-        if (sock[i] != PJ_INVALID_SOCKET)
-            pj_sock_close(sock[i]);
-    }
-    return status;
-}
-
 /* Check if sound device is idle. */
 static void check_snd_dev_idle()
@@ -579 +410 @@
      * there is no active call.
      */
-    if ((pjsua_var.snd_port!=NULL || pjsua_var.null_snd!=NULL) && 
+    if ((pjsua_var.snd_port!=NULL || pjsua_var.null_snd!=NULL) &&
         pjsua_var.snd_idle_timer.id == PJ_FALSE &&
         pjmedia_conf_get_connect_count(pjsua_var.mconf) == 0 &&
@@ -591 +422 @@
 
         pjsua_var.snd_idle_timer.id = PJ_TRUE;
-        pjsip_endpt_schedule_timer(pjsua_var.endpt, &pjsua_var.snd_idle_timer, 
+        pjsip_endpt_schedule_timer(pjsua_var.endpt, &pjsua_var.snd_idle_timer,
                                    &delay);
     }
@@ -605 +436 @@
     PJSUA_LOCK();
     if (entry->id) {
-        PJ_LOG(4,(THIS_FILE,"Closing sound device after idle for %d seconds", 
+        PJ_LOG(4,(THIS_FILE,"Closing sound device after idle for %d seconds",
                   pjsua_var.media_cfg.snd_auto_close_time));
 
@@ -623 +454 @@
     pj_status_t status;
 
+#if DISABLED_FOR_TICKET_1185
     /* Create media for calls, if none is specified */
-    if (pjsua_var.calls[0].med_tp == NULL) {
+    if (pjsua_var.calls[0].media[0].tp == NULL) {
         pjsua_transport_config transport_cfg;
 
@@ -635 +467 @@
             return status;
     }
-
-    pj_timer_entry_init(&pjsua_var.snd_idle_timer, PJ_FALSE, NULL, 
+#endif
+
+    pj_timer_entry_init(&pjsua_var.snd_idle_timer, PJ_FALSE, NULL,
                         &close_snd_timer_cb);
 
     /* Perform NAT detection */
-    pjsua_detect_nat_type();
+    status = pjsua_detect_nat_type();
+    if (status != PJ_SUCCESS) {
+        PJ_PERROR(1,(THIS_FILE, status, "NAT type detection failed"));
+    }
 
     return PJ_SUCCESS;
@@ -685 +521 @@
     /* Close media transports */
     for (i=0; i<pjsua_var.ua_cfg.max_calls; ++i) {
-        if (pjsua_var.calls[i].med_tp_st != PJSUA_MED_TP_IDLE) {
-            pjsua_media_channel_deinit(i);
-        }
-        if (pjsua_var.calls[i].med_tp && pjsua_var.calls[i].med_tp_auto_del) {
-            pjmedia_transport_close(pjsua_var.calls[i].med_tp);
-        }
-        pjsua_var.calls[i].med_tp = NULL;
+        unsigned strm_idx;
+        pjsua_call *call = &pjsua_var.calls[i];
+        for (strm_idx=0; strm_idx<call->med_cnt; ++strm_idx) {
+            pjsua_call_media *call_med = &call->media[strm_idx];
+            if (call_med->tp_st != PJSUA_MED_TP_IDLE) {
+                pjsua_media_channel_deinit(i);
+            }
+            if (call_med->tp && call_med->tp_auto_del) {
+                pjmedia_transport_close(call_med->tp);
+            }
+            call_med->tp = NULL;
+        }
     }
 
     /* Destroy media endpoint. */
     if (pjsua_var.med_endpt) {
+
+        /* Videodev */
+#       if PJMEDIA_HAS_VIDEO
+            pjmedia_vid_subsys_shutdown();
+#       endif
+
+        /* ffmpeg */
+#       if PJMEDIA_HAS_VIDEO && PJMEDIA_HAS_FFMPEG_CODEC
+            pjmedia_codec_ffmpeg_deinit();
+#       endif
 
         /* Shutdown all codecs: */
@@ -745 +596 @@
 }
 
-
+/*
+ * Create RTP and RTCP socket pair, and possibly resolve their public
+ * address via STUN.
+ */
+static pj_status_t create_rtp_rtcp_sock(const pjsua_transport_config *cfg,
+                                        pjmedia_sock_info *skinfo)
+{
+    enum {
+        RTP_RETRY = 100
+    };
+    int i;
+    pj_sockaddr_in bound_addr;
+    pj_sockaddr_in mapped_addr[2];
+    pj_status_t status = PJ_SUCCESS;
+    char addr_buf[PJ_INET6_ADDRSTRLEN+2];
+    pj_sock_t sock[2];
+
+    /* Make sure STUN server resolution has completed */
+    status = resolve_stun_server(PJ_TRUE);
+    if (status != PJ_SUCCESS) {
+        pjsua_perror(THIS_FILE, "Error resolving STUN server", status);
+        return status;
+    }
+
+    if (next_rtp_port == 0)
+        next_rtp_port = (pj_uint16_t)cfg->port;
+
+    if (next_rtp_port == 0)
+        next_rtp_port = (pj_uint16_t)40000;
+
+    for (i=0; i<2; ++i)
+        sock[i] = PJ_INVALID_SOCKET;
+
+    bound_addr.sin_addr.s_addr = PJ_INADDR_ANY;
+    if (cfg->bound_addr.slen) {
+        status = pj_sockaddr_in_set_str_addr(&bound_addr, &cfg->bound_addr);
+        if (status != PJ_SUCCESS) {
+            pjsua_perror(THIS_FILE, "Unable to resolve transport bind address",
+                         status);
+            return status;
+        }
+    }
+
+    /* Loop retry to bind RTP and RTCP sockets. */
+    for (i=0; i<RTP_RETRY; ++i, next_rtp_port += 2) {
+
+        /* Create RTP socket. */
+        status = pj_sock_socket(pj_AF_INET(), pj_SOCK_DGRAM(), 0, &sock[0]);
+        if (status != PJ_SUCCESS) {
+            pjsua_perror(THIS_FILE, "socket() error", status);
+            return status;
+        }
+
+        /* Apply QoS to RTP socket, if specified */
+        status = pj_sock_apply_qos2(sock[0], cfg->qos_type,
+                                    &cfg->qos_params,
+                                    2, THIS_FILE, "RTP socket");
+
+        /* Bind RTP socket */
+        status=pj_sock_bind_in(sock[0], pj_ntohl(bound_addr.sin_addr.s_addr),
+                               next_rtp_port);
+        if (status != PJ_SUCCESS) {
+            pj_sock_close(sock[0]);
+            sock[0] = PJ_INVALID_SOCKET;
+            continue;
+        }
+
+        /* Create RTCP socket. */
+        status = pj_sock_socket(pj_AF_INET(), pj_SOCK_DGRAM(), 0, &sock[1]);
+        if (status != PJ_SUCCESS) {
+            pjsua_perror(THIS_FILE, "socket() error", status);
+            pj_sock_close(sock[0]);
+            return status;
+        }
+
+        /* Apply QoS to RTCP socket, if specified */
+        status = pj_sock_apply_qos2(sock[1], cfg->qos_type,
+                                    &cfg->qos_params,
+                                    2, THIS_FILE, "RTCP socket");
+
+        /* Bind RTCP socket */
+        status=pj_sock_bind_in(sock[1], pj_ntohl(bound_addr.sin_addr.s_addr),
+                               (pj_uint16_t)(next_rtp_port+1));
+        if (status != PJ_SUCCESS) {
+            pj_sock_close(sock[0]);
+            sock[0] = PJ_INVALID_SOCKET;
+
+            pj_sock_close(sock[1]);
+            sock[1] = PJ_INVALID_SOCKET;
+            continue;
+        }
+
+        /*
+         * If we're configured to use STUN, then find out the mapped address,
+         * and make sure that the mapped RTCP port is adjacent with the RTP.
+         */
+        if (pjsua_var.stun_srv.addr.sa_family != 0) {
+            char ip_addr[32];
+            pj_str_t stun_srv;
+
+            pj_ansi_strcpy(ip_addr,
+                           pj_inet_ntoa(pjsua_var.stun_srv.ipv4.sin_addr));
+            stun_srv = pj_str(ip_addr);
+
+            status=pjstun_get_mapped_addr(&pjsua_var.cp.factory, 2, sock,
+                                           &stun_srv, pj_ntohs(pjsua_var.stun_srv.ipv4.sin_port),
+                                           &stun_srv, pj_ntohs(pjsua_var.stun_srv.ipv4.sin_port),
+                                           mapped_addr);
+            if (status != PJ_SUCCESS) {
+                pjsua_perror(THIS_FILE, "STUN resolve error", status);
+                goto on_error;
+            }
+
+#if PJSUA_REQUIRE_CONSECUTIVE_RTCP_PORT
+            if (pj_ntohs(mapped_addr[1].sin_port) ==
+                pj_ntohs(mapped_addr[0].sin_port)+1)
+            {
+                /* Success! */
+                break;
+            }
+
+            pj_sock_close(sock[0]);
+            sock[0] = PJ_INVALID_SOCKET;
+
+            pj_sock_close(sock[1]);
+            sock[1] = PJ_INVALID_SOCKET;
+#else
+            if (pj_ntohs(mapped_addr[1].sin_port) !=
+                pj_ntohs(mapped_addr[0].sin_port)+1)
+            {
+                PJ_LOG(4,(THIS_FILE,
+                          "Note: STUN mapped RTCP port %d is not adjacent"
+                          " to RTP port %d",
+                          pj_ntohs(mapped_addr[1].sin_port),
+                          pj_ntohs(mapped_addr[0].sin_port)));
+            }
+            /* Success! */
+            break;
+#endif
+
+        } else if (cfg->public_addr.slen) {
+
+            status = pj_sockaddr_in_init(&mapped_addr[0], &cfg->public_addr,
+                                         (pj_uint16_t)next_rtp_port);
+            if (status != PJ_SUCCESS)
+                goto on_error;
+
+            status = pj_sockaddr_in_init(&mapped_addr[1], &cfg->public_addr,
+                                         (pj_uint16_t)(next_rtp_port+1));
+            if (status != PJ_SUCCESS)
+                goto on_error;
+
+            break;
+
+        } else {
+
+            if (bound_addr.sin_addr.s_addr == 0) {
+                pj_sockaddr addr;
+
+                /* Get local IP address. */
+                status = pj_gethostip(pj_AF_INET(), &addr);
+                if (status != PJ_SUCCESS)
+                    goto on_error;
+
+                bound_addr.sin_addr.s_addr = addr.ipv4.sin_addr.s_addr;
+            }
+
+            for (i=0; i<2; ++i) {
+                pj_sockaddr_in_init(&mapped_addr[i], NULL, 0);
+                mapped_addr[i].sin_addr.s_addr = bound_addr.sin_addr.s_addr;
+            }
+
+            mapped_addr[0].sin_port=pj_htons((pj_uint16_t)next_rtp_port);
+            mapped_addr[1].sin_port=pj_htons((pj_uint16_t)(next_rtp_port+1));
+            break;
+        }
+    }
+
+    if (sock[0] == PJ_INVALID_SOCKET) {
+        PJ_LOG(1,(THIS_FILE,
+                  "Unable to find appropriate RTP/RTCP ports combination"));
+        goto on_error;
+    }
+
+
+    skinfo->rtp_sock = sock[0];
+    pj_memcpy(&skinfo->rtp_addr_name,
+              &mapped_addr[0], sizeof(pj_sockaddr_in));
+
+    skinfo->rtcp_sock = sock[1];
+    pj_memcpy(&skinfo->rtcp_addr_name,
+              &mapped_addr[1], sizeof(pj_sockaddr_in));
+
+    PJ_LOG(4,(THIS_FILE, "RTP socket reachable at %s",
+              pj_sockaddr_print(&skinfo->rtp_addr_name, addr_buf,
+                                sizeof(addr_buf), 3)));
+    PJ_LOG(4,(THIS_FILE, "RTCP socket reachable at %s",
+              pj_sockaddr_print(&skinfo->rtcp_addr_name, addr_buf,
+                                sizeof(addr_buf), 3)));
+
+    next_rtp_port += 2;
+    return PJ_SUCCESS;
+
+on_error:
+    for (i=0; i<2; ++i) {
+        if (sock[i] != PJ_INVALID_SOCKET)
+            pj_sock_close(sock[i]);
+    }
+    return status;
+}
+
+/* Create normal UDP media transports */
+static pj_status_t create_udp_media_transport(const pjsua_transport_config *cfg,
+                                              pjsua_call_media *call_med)
+{
+    pjmedia_sock_info skinfo;
+    pj_status_t status;
+
+    status = create_rtp_rtcp_sock(cfg, &skinfo);
+    if (status != PJ_SUCCESS) {
+        pjsua_perror(THIS_FILE, "Unable to create RTP/RTCP socket",
+                     status);
+        goto on_error;
+    }
+
+    status = pjmedia_transport_udp_attach(pjsua_var.med_endpt, NULL,
+                                          &skinfo, 0, &call_med->tp);
+    if (status != PJ_SUCCESS) {
+        pjsua_perror(THIS_FILE, "Unable to create media transport",
+                     status);
+        goto on_error;
+    }
+
+    pjmedia_transport_simulate_lost(call_med->tp, PJMEDIA_DIR_ENCODING,
+                                    pjsua_var.media_cfg.tx_drop_pct);
+
+    pjmedia_transport_simulate_lost(call_med->tp, PJMEDIA_DIR_DECODING,
+                                    pjsua_var.media_cfg.rx_drop_pct);
+
+    return PJ_SUCCESS;
+
+on_error:
+    if (call_med->tp)
+        pjmedia_transport_close(call_med->tp);
+
+    return status;
+}
+
+#if DISABLED_FOR_TICKET_1185
 /* Create normal UDP media transports */
 static pj_status_t create_udp_media_transports(pjsua_transport_config *cfg)
 {
     unsigned i;
-    pjmedia_sock_info skinfo;
     pj_status_t status;
 
-    /* Create each media transport */
-    for (i=0; i<pjsua_var.ua_cfg.max_calls; ++i) {
-
-        status = create_rtp_rtcp_sock(cfg, &skinfo);
-        if (status != PJ_SUCCESS) {
-            pjsua_perror(THIS_FILE, "Unable to create RTP/RTCP socket",
-                         status);
-            goto on_error;
-        }
-
-        status = pjmedia_transport_udp_attach(pjsua_var.med_endpt, NULL,
-                                              &skinfo, 0,
-                                              &pjsua_var.calls[i].med_tp);
-        if (status != PJ_SUCCESS) {
-            pjsua_perror(THIS_FILE, "Unable to create media transport",
-                         status);
-            goto on_error;
-        }
-
-        pjmedia_transport_simulate_lost(pjsua_var.calls[i].med_tp,
-                                        PJMEDIA_DIR_ENCODING,
-                                        pjsua_var.media_cfg.tx_drop_pct);
-
-        pjmedia_transport_simulate_lost(pjsua_var.calls[i].med_tp,
-                                        PJMEDIA_DIR_DECODING,
-                                        pjsua_var.media_cfg.rx_drop_pct);
-
+    for (i=0; i < pjsua_var.ua_cfg.max_calls; ++i) {
+        pjsua_call *call = &pjsua_var.calls[i];
+        unsigned strm_idx;
+
+        for (strm_idx=0; strm_idx < call->med_cnt; ++strm_idx) {
+            pjsua_call_media *call_med = &call->media[strm_idx];
+
+            status = create_udp_media_transport(cfg, &call_med->tp);
+            if (status != PJ_SUCCESS)
+                goto on_error;
+        }
     }
 
@@ -785 +867 @@
 
 on_error:
-    for (i=0; i<pjsua_var.ua_cfg.max_calls; ++i) {
-        if (pjsua_var.calls[i].med_tp != NULL) {
-            pjmedia_transport_close(pjsua_var.calls[i].med_tp);
-            pjsua_var.calls[i].med_tp = NULL;
-        }
-    }
-
+    for (i=0; i < pjsua_var.ua_cfg.max_calls; ++i) {
+        pjsua_call *call = &pjsua_var.calls[i];
+        unsigned strm_idx;
+
+        for (strm_idx=0; strm_idx < call->med_cnt; ++strm_idx) {
+            pjsua_call_media *call_med = &call->media[strm_idx];
+
+            if (call_med->tp) {
+                pjmedia_transport_close(call_med->tp);
+                call_med->tp = NULL;
+            }
+        }
+    }
     return status;
 }
-
+#endif
 
 /* This callback is called when ICE negotiation completes */
@@ -801 +889 @@
                             pj_status_t result)
 {
-    unsigned id;
-    pj_bool_t found = PJ_FALSE;
-
-    /* Find call which has this media transport */
-
-    PJSUA_LOCK();
-
-    for (id=0; id<pjsua_var.ua_cfg.max_calls; ++id) {
-        if (pjsua_var.calls[id].med_tp == tp ||
-            pjsua_var.calls[id].med_orig == tp) 
-        {
-            found = PJ_TRUE;
-            break;
-        }
-    }
-
-    PJSUA_UNLOCK();
-
-    if (!found)
+    pjsua_call_media *call_med = (pjsua_call_media*)tp->user_data;
+
+    if (!call_med)
         return;
 
     switch (op) {
     case PJ_ICE_STRANS_OP_INIT:
-        pjsua_var.calls[id].med_tp_ready = result;
+        call_med->tp_ready = result;
         break;
     case PJ_ICE_STRANS_OP_NEGOTIATION:
         if (result != PJ_SUCCESS) {
-            pjsua_var.calls[id].media_st = PJSUA_CALL_MEDIA_ERROR;
-            pjsua_var.calls[id].media_dir = PJMEDIA_DIR_NONE;
-
-            if (pjsua_var.ua_cfg.cb.on_call_media_state) {
-                pjsua_var.ua_cfg.cb.on_call_media_state(id);
+            call_med->state = PJSUA_CALL_MEDIA_ERROR;
+            call_med->dir = PJMEDIA_DIR_NONE;
+
+            if (call_med->call && pjsua_var.ua_cfg.cb.on_call_media_state) {
+                pjsua_var.ua_cfg.cb.on_call_media_state(call_med->call->index);
             }
-        } else {
+        } else if (call_med->call) {
             /* Send UPDATE if default transport address is different than
              * what was advertised (ticket #881)
@@ -854 +926 @@
             if (ii && ii->role==PJ_ICE_SESS_ROLE_CONTROLLING &&
                 pj_sockaddr_cmp(&tpinfo.sock_info.rtp_addr_name,
-                                &pjsua_var.calls[id].med_rtp_addr))
+                                &call_med->rtp_addr))
             {
                 pj_bool_t use_update;
@@ -861 +933 @@
                 pjsip_dialog *dlg;
 
-                dlg = pjsua_var.calls[id].inv->dlg;
+                dlg = call_med->call->inv->dlg;
                 support_update = pjsip_dlg_remote_has_cap(dlg, PJSIP_H_ALLOW,
                                                           NULL, &STR_UPDATE);
@@ -868 +940 @@
                 PJ_LOG(4,(THIS_FILE, 
                           "ICE default transport address has changed for "
-                          "call %d, sending %s", id,
+                          "call %d, sending %s", call_med->call->index,
                           (use_update ? "UPDATE" : "re-INVITE")));
 
                 if (use_update)
-                    pjsua_call_update(id, 0, NULL);
+                    pjsua_call_update(call_med->call->index, 0, NULL);
                 else
-                    pjsua_call_reinvite(id, 0, NULL);
+                    pjsua_call_reinvite(call_med->call->index, 0, NULL);
             }
         }
@@ -881 +953 @@
         if (result != PJ_SUCCESS) {
             PJ_PERROR(4,(THIS_FILE, result,
-                         "ICE keep alive failure for transport %d", id));
+                         "ICE keep alive failure for transport %d:%d",
+                         call_med->call->index, call_med->idx));
         }
         if (pjsua_var.ua_cfg.cb.on_ice_transport_error) {
+            pjsua_call_id id = call_med->call->index;
             (*pjsua_var.ua_cfg.cb.on_ice_transport_error)(id, op, result,
                                                           NULL);
@@ -919 +993 @@
 
 /* Create ICE media transports (when ice is enabled) */
-static pj_status_t create_ice_media_transports(pjsua_transport_config *cfg)
+static pj_status_t create_ice_media_transport(
+                                const pjsua_transport_config *cfg,
+                                pjsua_call_media *call_med)
 {
     char stunip[PJ_INET6_ADDRSTRLEN];
     pj_ice_strans_cfg ice_cfg;
-    unsigned i;
+    pjmedia_ice_cb ice_cb;
+    char name[32];
+    unsigned comp_cnt;
     pj_status_t status;
 
@@ -980 +1058 @@
     }
 
-    /* Create each media transport */
-    for (i=0; i<pjsua_var.ua_cfg.max_calls; ++i) {
-        pjmedia_ice_cb ice_cb;
-        char name[32];
-        unsigned comp_cnt;
-
-        pj_bzero(&ice_cb, sizeof(pjmedia_ice_cb));
-        ice_cb.on_ice_complete = &on_ice_complete;
-        pj_ansi_snprintf(name, sizeof(name), "icetp%02d", i);
-        pjsua_var.calls[i].med_tp_ready = PJ_EPENDING;
-
-        comp_cnt = 1;
-        if (PJMEDIA_ADVERTISE_RTCP && !pjsua_var.media_cfg.ice_no_rtcp)
-            ++comp_cnt;
-
-        status = pjmedia_ice_create(pjsua_var.med_endpt, name, comp_cnt,
-                                    &ice_cfg, &ice_cb,
-                                    &pjsua_var.calls[i].med_tp);
-        if (status != PJ_SUCCESS) {
-            pjsua_perror(THIS_FILE, "Unable to create ICE media transport",
-                         status);
-            goto on_error;
-        }
-
-        /* Wait until transport is initialized, or time out */
-        PJSUA_UNLOCK();
-        while (pjsua_var.calls[i].med_tp_ready == PJ_EPENDING) {
-            pjsua_handle_events(100);
-        }
-        PJSUA_LOCK();
-        if (pjsua_var.calls[i].med_tp_ready != PJ_SUCCESS) {
-            pjsua_perror(THIS_FILE, "Error initializing ICE media transport",
-                         pjsua_var.calls[i].med_tp_ready);
-            status = pjsua_var.calls[i].med_tp_ready;
-            goto on_error;
-        }
-
-        pjmedia_transport_simulate_lost(pjsua_var.calls[i].med_tp,
-                                        PJMEDIA_DIR_ENCODING,
-                                        pjsua_var.media_cfg.tx_drop_pct);
-
-        pjmedia_transport_simulate_lost(pjsua_var.calls[i].med_tp,
-                                        PJMEDIA_DIR_DECODING,
-                                        pjsua_var.media_cfg.rx_drop_pct);
-    }
+    pj_bzero(&ice_cb, sizeof(pjmedia_ice_cb));
+    ice_cb.on_ice_complete = &on_ice_complete;
+    pj_ansi_snprintf(name, sizeof(name), "icetp%02d", call_med->idx);
+    call_med->tp_ready = PJ_EPENDING;
+
+    comp_cnt = 1;
+    if (PJMEDIA_ADVERTISE_RTCP && !pjsua_var.media_cfg.ice_no_rtcp)
+        ++comp_cnt;
+
+    status = pjmedia_ice_create(pjsua_var.med_endpt, name, comp_cnt,
+                                &ice_cfg, &ice_cb, &call_med->tp);
+    if (status != PJ_SUCCESS) {
+        pjsua_perror(THIS_FILE, "Unable to create ICE media transport",
+                     status);
+        goto on_error;
+    }
+
+    /* Wait until transport is initialized, or time out */
+    PJSUA_UNLOCK();
+    while (call_med->tp_ready == PJ_EPENDING) {
+        pjsua_handle_events(100);
+    }
+    PJSUA_LOCK();
+    if (call_med->tp_ready != PJ_SUCCESS) {
+        pjsua_perror(THIS_FILE, "Error initializing ICE media transport",
+                     call_med->tp_ready);
+        status = call_med->tp_ready;
+        goto on_error;
+    }
+
+    pjmedia_transport_simulate_lost(call_med->tp, PJMEDIA_DIR_ENCODING,
+                                    pjsua_var.media_cfg.tx_drop_pct);
+
+    pjmedia_transport_simulate_lost(call_med->tp, PJMEDIA_DIR_DECODING,
+                                    pjsua_var.media_cfg.rx_drop_pct);
 
     return PJ_SUCCESS;
 
 on_error:
-    for (i=0; i<pjsua_var.ua_cfg.max_calls; ++i) {
-        if (pjsua_var.calls[i].med_tp != NULL) {
-            pjmedia_transport_close(pjsua_var.calls[i].med_tp);
-            pjsua_var.calls[i].med_tp = NULL;
-        }
+    if (call_med->tp != NULL) {
+        pjmedia_transport_close(call_med->tp);
+        call_med->tp = NULL;
     }
 
@@ -1039 +1105 @@
 }
 
-
-/*
- * Create UDP media transports for all the calls. This function creates
+#if DISABLED_FOR_TICKET_1185
+/* Create ICE media transports (when ice is enabled) */
+static pj_status_t create_ice_media_transports(pjsua_transport_config *cfg)
+{
+    unsigned i;
+    pj_status_t status;
+
+    for (i=0; i < pjsua_var.ua_cfg.max_calls; ++i) {
+        pjsua_call *call = &pjsua_var.calls[i];
+        unsigned strm_idx;
+
+        for (strm_idx=0; strm_idx < call->med_cnt; ++strm_idx) {
+            pjsua_call_media *call_med = &call->media[strm_idx];
+
+            status = create_ice_media_transport(cfg, call_med);
+            if (status != PJ_SUCCESS)
+                goto on_error;
+        }
+    }
+
+    return PJ_SUCCESS;
+
+on_error:
+    for (i=0; i < pjsua_var.ua_cfg.max_calls; ++i) {
+        pjsua_call *call = &pjsua_var.calls[i];
+        unsigned strm_idx;
+
+        for (strm_idx=0; strm_idx < call->med_cnt; ++strm_idx) {
+            pjsua_call_media *call_med = &call->media[strm_idx];
+
+            if (call_med->tp) {
+                pjmedia_transport_close(call_med->tp);
+                call_med->tp = NULL;
+            }
+        }
+    }
+    return status;
+}
+#endif
+
+#if DISABLED_FOR_TICKET_1185
+/*
+ * Create media transports for all the calls. This function creates
  * one UDP media transport for each call.
  */
@@ -1059 +1165 @@
     /* Delete existing media transports */
     for (i=0; i<pjsua_var.ua_cfg.max_calls; ++i) {
-        if (pjsua_var.calls[i].med_tp != NULL && 
-            pjsua_var.calls[i].med_tp_auto_del) 
-        {
-            pjmedia_transport_close(pjsua_var.calls[i].med_tp);
-            pjsua_var.calls[i].med_tp = NULL;
-            pjsua_var.calls[i].med_orig = NULL;
+        pjsua_call *call = &pjsua_var.calls[i];
+        unsigned strm_idx;
+
+        for (strm_idx=0; strm_idx < call->med_cnt; ++strm_idx) {
+            pjsua_call_media *call_med = &call->media[strm_idx];
+
+            if (call_med->tp && call_med->tp_auto_del) {
+                pjmedia_transport_close(call_med->tp);
+                call_med->tp = NULL;
+                call_med->tp_orig = NULL;
+            }
         }
     }
@@ -1080 +1191 @@
     /* Set media transport auto_delete to True */
     for (i=0; i<pjsua_var.ua_cfg.max_calls; ++i) {
-        pjsua_var.calls[i].med_tp_auto_del = PJ_TRUE;
+        pjsua_call *call = &pjsua_var.calls[i];
+        unsigned strm_idx;
+
+        for (strm_idx=0; strm_idx < call->med_cnt; ++strm_idx) {
+            pjsua_call_media *call_med = &call->media[strm_idx];
+
+            call_med->tp_auto_del = PJ_TRUE;
+        }
     }
 
@@ -1101 +1219 @@
     /* Assign the media transports */
     for (i=0; i<pjsua_var.ua_cfg.max_calls; ++i) {
-        if (pjsua_var.calls[i].med_tp != NULL && 
-            pjsua_var.calls[i].med_tp_auto_del) 
-        {
-            pjmedia_transport_close(pjsua_var.calls[i].med_tp);
-        }
-
-        pjsua_var.calls[i].med_tp = tp[i].transport;
-        pjsua_var.calls[i].med_tp_auto_del = auto_delete;
+        pjsua_call *call = &pjsua_var.calls[i];
+        unsigned strm_idx;
+
+        for (strm_idx=0; strm_idx < call->med_cnt; ++strm_idx) {
+            pjsua_call_media *call_med = &call->media[strm_idx];
+
+            if (call_med->tp && call_med->tp_auto_del) {
+                pjmedia_transport_close(call_med->tp);
+                call_med->tp = NULL;
+                call_med->tp_orig = NULL;
+            }
+        }
+
+        PJ_TODO(remove_pjsua_media_transports_attach);
+
+        call->media[0].tp = tp[i].transport;
+        call->media[0].tp_auto_del = auto_delete;
     }
 
     return PJ_SUCCESS;
 }
-
-
-static int find_audio_index(const pjmedia_sdp_session *sdp, 
-                            pj_bool_t prefer_srtp)
+#endif
+
+/* Go through the list of media in the SDP, find acceptable media, and
+ * sort them based on the "quality" of the media, and store the indexes
+ * in the specified array. Media with the best quality will be listed
+ * first in the array. The quality factors considered currently is
+ * encryption.
+ */
+static void sort_media(const pjmedia_sdp_session *sdp,
+                       const pj_str_t *type,
+                       pjmedia_srtp_use use_srtp,
+                       pj_uint8_t midx[],
+                       unsigned *p_count)
 {
     unsigned i;
-    int audio_idx = -1;
-
-    for (i=0; i<sdp->media_count; ++i) {
+    unsigned count = 0;
+    int score[PJSUA_MAX_CALL_MEDIA];
+
+    pj_assert(*p_count >= PJSUA_MAX_CALL_MEDIA);
+
+    *p_count = 0;
+    pj_bzero(score, sizeof(score));
+
+    /* Score each media */
+    for (i=0; i<sdp->media_count && count<PJSUA_MAX_CALL_MEDIA; ++i) {
         const pjmedia_sdp_media *m = sdp->media[i];
 
-        /* Skip if media is not audio */
-        if (pj_stricmp2(&m->desc.media, "audio") != 0)
+        /* Skip different media */
+        if (pj_stricmp(&m->desc.media, type) != 0) {
+            score[count++] = -22000;
             continue;
-
-        /* Skip if media is disabled */
+        }
+
+        /* Supported transports */
+        if (pj_stricmp2(&m->desc.transport, "RTP/SAVP")==0) {
+            switch (use_srtp) {
+            case PJMEDIA_SRTP_MANDATORY:
+            case PJMEDIA_SRTP_OPTIONAL:
+                ++score[i];
+                break;
+            case PJMEDIA_SRTP_DISABLED:
+                --score[i];
+                break;
+            }
+        } else if (pj_stricmp2(&m->desc.transport, "RTP/AVP")==0) {
+            switch (use_srtp) {
+            case PJMEDIA_SRTP_MANDATORY:
+                --score[i];
+                break;
+            case PJMEDIA_SRTP_OPTIONAL:
+                /* No change in score */
+                break;
+            case PJMEDIA_SRTP_DISABLED:
+                ++score[i];
+                break;
+            }
+        } else {
+            score[i] -= 10;
+        }
+
+        /* Is media disabled? */
         if (m->desc.port == 0)
-            continue;
-
-        /* Skip if transport is not supported */
-        if (pj_stricmp2(&m->desc.transport, "RTP/AVP") != 0 &&
-            pj_stricmp2(&m->desc.transport, "RTP/SAVP") != 0)
-        {
-            continue;
-        }
-
-        if (audio_idx == -1) {
-            audio_idx = i;
+            score[i] -= 10;
+
+        /* Is media inactive? */
+        if (pjmedia_sdp_media_find_attr2(m, "inactive", NULL))
+            score[i] -= 10;
+
+        ++count;
+    }
+
+    /* Created sorted list based on quality */
+    for (i=0; i<count; ++i) {
+        unsigned j;
+        int best = 0;
+
+        for (j=1; j<count; ++j) {
+            if (score[j] > score[best])
+                best = j;
+        }
+        /* Don't put media with negative score, that media is unacceptable
+         * for us.
+         */
+        if (score[best] >= 0) {
+            midx[*p_count] = (pj_uint8_t)best;
+            (*p_count)++;
+        }
+
+        score[best] = -22000;
+
+    }
+}
+
+/* Initialize the media line */
+static pj_status_t pjsua_call_media_init(pjsua_call_media *call_med,
+                                         pjmedia_type type,
+                                         const pjsua_transport_config *tcfg,
+                                         int security_level,
+                                         int *sip_err_code)
+{
+    pjsua_acc *acc = &pjsua_var.acc[call_med->call->acc_id];
+    pj_status_t status;
+
+    /*
+     * Note: this function may be called when the media already exists
+     * (e.g. in reinvites, updates, etc.)
+     */
+    call_med->type = type;
+
+    /* Create the media transport for initial call. This is blocking for now */
+    if (call_med->tp == NULL) {
+        if (pjsua_var.media_cfg.enable_ice) {
+            status = create_ice_media_transport(tcfg, call_med);
         } else {
-            /* We've found multiple candidates. This could happen
-             * e.g. when remote is offering both RTP/SAVP and RTP/AVP,
-             * or when remote for some reason offers two audio.
-             */
-
-            if (prefer_srtp &&
-                pj_stricmp2(&m->desc.transport, "RTP/SAVP")==0)
-            {
-                /* Prefer RTP/SAVP when our media transport is SRTP */
-                audio_idx = i;
-                break;
-            } else if (!prefer_srtp &&
-                       pj_stricmp2(&m->desc.transport, "RTP/AVP")==0)
-            {
-                /* Prefer RTP/AVP when our media transport is NOT SRTP */
-                audio_idx = i;
+            status = create_udp_media_transport(tcfg, call_med);
+        }
+
+        if (status != PJ_SUCCESS) {
+            PJ_PERROR(1,(THIS_FILE, status, "Error creating media transport"));
+            return status;
+        }
+    }
+
+#if defined(PJMEDIA_HAS_SRTP) && (PJMEDIA_HAS_SRTP != 0)
+    /* This function may be called when SRTP transport already exists
+     * (e.g: in re-invite, update), don't need to destroy/re-create.
+     */
+    if (!call_med->tp_orig || call_med->tp == call_med->tp_orig) {
+        pjmedia_srtp_setting srtp_opt;
+        pjmedia_transport *srtp = NULL;
+
+        /* Check if SRTP requires secure signaling */
+        if (acc->cfg.use_srtp != PJMEDIA_SRTP_DISABLED) {
+            if (security_level < acc->cfg.srtp_secure_signaling) {
+                if (sip_err_code)
+                    *sip_err_code = PJSIP_SC_NOT_ACCEPTABLE;
+                status = PJSIP_ESESSIONINSECURE;
+                goto on_error;
             }
         }
-    }
-
-    return audio_idx;
-}
-
+
+        /* Always create SRTP adapter */
+        pjmedia_srtp_setting_default(&srtp_opt);
+        srtp_opt.close_member_tp = PJ_FALSE;
+        /* If media session has been ever established, let's use remote's
+         * preference in SRTP usage policy, especially when it is stricter.
+         */
+        if (call_med->rem_srtp_use > acc->cfg.use_srtp)
+            srtp_opt.use = call_med->rem_srtp_use;
+        else
+            srtp_opt.use = acc->cfg.use_srtp;
+
+        status = pjmedia_transport_srtp_create(pjsua_var.med_endpt,
+                                               call_med->tp,
+                                               &srtp_opt, &srtp);
+        if (status != PJ_SUCCESS) {
+            if (sip_err_code)
+                *sip_err_code = PJSIP_SC_INTERNAL_SERVER_ERROR;
+            goto on_error;
+        }
+
+        /* Set SRTP as current media transport */
+        call_med->tp_orig = call_med->tp;
+        call_med->tp = srtp;
+    }
+#else
+    call->tp_orig = call->tp;
+    PJ_UNUSED_ARG(security_level);
+#endif
+
+    return PJ_SUCCESS;
+
+on_error:
+    if (call_med->tp) {
+        pjmedia_transport_close(call_med->tp);
+        call_med->tp = NULL;
+    }
+    return status;
+}
 
 pj_status_t pjsua_media_channel_init(pjsua_call_id call_id,
@@ -1173 +1427 @@
                                      int *sip_err_code)
 {
+    const pj_str_t STR_AUDIO = { "audio", 5 };
+    const pj_str_t STR_VIDEO = { "video", 5 };
     pjsua_call *call = &pjsua_var.calls[call_id];
+    pjsua_acc *acc = &pjsua_var.acc[call->acc_id];
+    pj_uint8_t maudidx[PJSUA_MAX_CALL_MEDIA];
+    unsigned maudcnt = PJ_ARRAY_SIZE(maudidx);
+    pj_uint8_t mvididx[PJSUA_MAX_CALL_MEDIA];
+    unsigned mvidcnt = PJ_ARRAY_SIZE(mvididx);
+    pjmedia_type media_types[PJSUA_MAX_CALL_MEDIA];
+    unsigned mi;
     pj_status_t status;
 
-#if defined(PJMEDIA_HAS_SRTP) && (PJMEDIA_HAS_SRTP != 0)
-    pjsua_acc *acc = &pjsua_var.acc[call->acc_id];
-    pjmedia_srtp_setting srtp_opt;
-    pjmedia_transport *srtp = NULL;
-#endif
-
     PJ_UNUSED_ARG(role);
 
+    if (pjsua_get_state() != PJSUA_STATE_RUNNING)
+        return PJ_EBUSY;
+
+#if DISABLED_FOR_TICKET_1185
     /* Return error if media transport has not been created yet
      * (e.g. application is starting)
      */
-    if (call->med_tp == NULL) {
-        if (sip_err_code)
-            *sip_err_code = PJSIP_SC_INTERNAL_SERVER_ERROR;
-        return PJ_EBUSY;
-    }
-
-#if defined(PJMEDIA_HAS_SRTP) && (PJMEDIA_HAS_SRTP != 0)
-    /* This function may be called when SRTP transport already exists 
-     * (e.g: in re-invite, update), don't need to destroy/re-create.
-     */
-    if (!call->med_orig || call->med_tp == call->med_orig) {
-
-        /* Check if SRTP requires secure signaling */
-        if (acc->cfg.use_srtp != PJMEDIA_SRTP_DISABLED) {
-            if (security_level < acc->cfg.srtp_secure_signaling) {
-                if (sip_err_code)
-                    *sip_err_code = PJSIP_SC_NOT_ACCEPTABLE;
-                return PJSIP_ESESSIONINSECURE;
-            }
-        }
-
-        /* Always create SRTP adapter */
-        pjmedia_srtp_setting_default(&srtp_opt);
-        srtp_opt.close_member_tp = PJ_FALSE;
-        /* If media session has been ever established, let's use remote's 
-         * preference in SRTP usage policy, especially when it is stricter.
-         */
-        if (call->rem_srtp_use > acc->cfg.use_srtp)
-            srtp_opt.use = call->rem_srtp_use;
-        else
-            srtp_opt.use = acc->cfg.use_srtp;
-
-        status = pjmedia_transport_srtp_create(pjsua_var.med_endpt, 
-                                               call->med_tp,
-                                               &srtp_opt, &srtp);
-        if (status != PJ_SUCCESS) {
-            if (sip_err_code)
-                *sip_err_code = PJSIP_SC_INTERNAL_SERVER_ERROR;
-            return status;
-        }
-
-        /* Set SRTP as current media transport */
-        call->med_orig = call->med_tp;
-        call->med_tp = srtp;
-    }
-#else
-    call->med_orig = call->med_tp;
-    PJ_UNUSED_ARG(security_level);
+    for (i=0; i<call->med_cnt; ++i) {
+        if (call->media[i].tp == NULL) {
+            return PJ_EBUSY;
+        }
+    }
 #endif
 
-    /* Find out which media line in SDP that we support. If we are offerer,
-     * audio will be initialized at index 0 in SDP. 
-     */
-    if (rem_sdp == NULL) {
-        call->audio_idx = 0;
-    } 
-    /* Otherwise find out the candidate audio media line in SDP */
-    else {
-        pj_bool_t srtp_active;
-
-#if defined(PJMEDIA_HAS_SRTP) && (PJMEDIA_HAS_SRTP != 0)
-        srtp_active = acc->cfg.use_srtp;
-#else
-        srtp_active = PJ_FALSE;
-#endif
-
-        /* Media count must have been checked */
-        pj_assert(rem_sdp->media_count != 0);
-
-        call->audio_idx = find_audio_index(rem_sdp, srtp_active);
-    }
-
-    /* Reject offer if we couldn't find a good m=audio line in offer */
-    if (call->audio_idx < 0) {
+    if (rem_sdp) {
+        sort_media(rem_sdp, &STR_AUDIO, acc->cfg.use_srtp,
+                   maudidx, &maudcnt);
+
+        if (maudcnt==0) {
+            /* Expecting audio in the offer */
+            if (sip_err_code) *sip_err_code = PJSIP_SC_NOT_ACCEPTABLE_HERE;
+            pjsua_media_channel_deinit(call_id);
+            return PJSIP_ERRNO_FROM_SIP_STATUS(PJSIP_SC_NOT_ACCEPTABLE_HERE);
+        }
+
+        sort_media(rem_sdp, &STR_VIDEO, acc->cfg.use_srtp,
+                   mvididx, &mvidcnt);
+        mvidcnt = (mvidcnt < acc->cfg.max_video_cnt) ?
+                        mvidcnt : acc->cfg.max_video_cnt;
+
+    } else {
+        maudcnt = acc->cfg.max_audio_cnt;
+        for (mi=0; mi<maudcnt; ++mi) {
+            maudidx[mi] = mi;
+            media_types[mi] = PJMEDIA_TYPE_AUDIO;
+        }
+        mvidcnt = acc->cfg.max_video_cnt;
+        for (mi=0; mi<mvidcnt; ++mi) {
+            media_types[maudcnt + mi] = PJMEDIA_TYPE_VIDEO;
+        }
+    }
+
+    call->med_cnt = maudcnt + mvidcnt;
+
+    if (call->med_cnt == 0) {
+        /* Expecting at least one media */
         if (sip_err_code) *sip_err_code = PJSIP_SC_NOT_ACCEPTABLE_HERE;
         pjsua_media_channel_deinit(call_id);
@@ -1266 +1492 @@
     }
 
-    PJ_LOG(4,(THIS_FILE, "Media index %d selected for call %d",
+    /* Initialize each media line */
+    for (mi=0; mi < call->med_cnt; ++mi) {
+        pjsua_call_media *call_med = &call->media[mi];
+        pj_bool_t enabled = PJ_FALSE;
+        pjmedia_type media_type = PJMEDIA_TYPE_NONE;
+
+        if (rem_sdp) {
+            if (!pj_stricmp(&rem_sdp->media[mi]->desc.media, &STR_AUDIO)) {
+                media_type = PJMEDIA_TYPE_AUDIO;
+                if (pj_memchr(maudidx, mi, maudcnt * sizeof(maudidx[0]))) {
+                    enabled = PJ_TRUE;
+                }
+            }
+            else if (!pj_stricmp(&rem_sdp->media[mi]->desc.media, &STR_VIDEO)) {
+                media_type = PJMEDIA_TYPE_VIDEO;
+                if (pj_memchr(mvididx, mi, mvidcnt * sizeof(mvididx[0]))) {
+                    enabled = PJ_TRUE;
+                }
+            }
+
+        } else {
+            enabled = PJ_TRUE;
+            media_type = media_types[mi];
+        }
+
+        if (enabled) {
+            status = pjsua_call_media_init(call_med, media_type,
+                                           &acc->cfg.rtp_cfg,
+                                           security_level, sip_err_code);
+            if (status != PJ_SUCCESS) {
+                pjsua_media_channel_deinit(call_id);
+                return status;
+            }
+        } else {
+            /* By convention, the media is inactive if transport is NULL */
+            if (call_med->tp) {
+                pjmedia_transport_close(call_med->tp);
+                call_med->tp = NULL;
+            }
+        }
+    }
+
+    call->audio_idx = maudidx[0];
+
+    PJ_LOG(4,(THIS_FILE, "Media index %d selected for audio call %d",
               call->audio_idx, call->index));
 
-    /* Create the media transport */
-    status = pjmedia_transport_media_create(call->med_tp, tmp_pool, 0,
-                                            rem_sdp, call->audio_idx);
-    if (status != PJ_SUCCESS) {
-        if (sip_err_code) *sip_err_code = PJSIP_SC_NOT_ACCEPTABLE;
-        pjsua_media_channel_deinit(call_id);
-        return status;
-    }
-
-    call->med_tp_st = PJSUA_MED_TP_INIT;
+    /* Tell the media transport of a new offer/answer session */
+    for (mi=0; mi < call->med_cnt; ++mi) {
+        pjsua_call_media *call_med = &call->media[mi];
+
+        /* Note: tp may be NULL if this media line is inactive */
+        if (call_med->tp) {
+            status = pjmedia_transport_media_create(call_med->tp,
+                                                    tmp_pool, 0,
+                                                    rem_sdp, mi);
+            if (status != PJ_SUCCESS) {
+                if (sip_err_code) *sip_err_code = PJSIP_SC_NOT_ACCEPTABLE;
+                pjsua_media_channel_deinit(call_id);
+                return status;
+            }
+
+            call_med->tp_st = PJSUA_MED_TP_INIT;
+        }
+    }
+
     return PJ_SUCCESS;
 }
@@ -1286 +1565 @@
                                            const pjmedia_sdp_session *rem_sdp,
                                            pjmedia_sdp_session **p_sdp,
-                                           int *sip_status_code)
-{
-    enum { MAX_MEDIA = 1 };
+                                           int *sip_err_code)
+{
+    const pj_str_t STR_AUDIO = { "audio", 5 };
+    enum { MAX_MEDIA = PJSUA_MAX_CALL_MEDIA };
     pjmedia_sdp_session *sdp;
-    pjmedia_transport_info tpinfo;
+    pj_sockaddr origin;
     pjsua_call *call = &pjsua_var.calls[call_id];
+    pjsua_acc *acc = &pjsua_var.acc[call->acc_id];
     pjmedia_sdp_neg_state sdp_neg_state = PJMEDIA_SDP_NEG_STATE_NULL;
+    unsigned mi;
     pj_status_t status;
 
-    /* Return error if media transport has not been created yet
-     * (e.g. application is starting)
+    if (pjsua_get_state() != PJSUA_STATE_RUNNING)
+        return PJ_EBUSY;
+
+    if (rem_sdp) {
+        pj_uint8_t maudidx[PJSUA_MAX_CALL_MEDIA];
+        unsigned maudcnt = PJ_ARRAY_SIZE(maudidx);
+
+        sort_media(rem_sdp, &STR_AUDIO, acc->cfg.use_srtp,
+                   maudidx, &maudcnt);
+
+        if (maudcnt==0) {
+            /* Expecting audio in the offer */
+            if (sip_err_code) *sip_err_code = PJSIP_SC_NOT_ACCEPTABLE_HERE;
+            pjsua_media_channel_deinit(call_id);
+            return PJSIP_ERRNO_FROM_SIP_STATUS(PJSIP_SC_NOT_ACCEPTABLE_HERE);
+        }
+
+        call->audio_idx = maudidx[0];
+    } else {
+        /* Audio is first in our offer, by convention */
+        call->audio_idx = 0;
+    }
+
+    /* Create media if it's not created. This could happen when call is
+     * currently on-hold (with the old style hold)
      */
-    if (call->med_tp == NULL) {
-        return PJ_EBUSY;
-    }
-
-    if (rem_sdp && rem_sdp->media_count != 0) {
-        pj_bool_t srtp_active;
-
-#if defined(PJMEDIA_HAS_SRTP) && (PJMEDIA_HAS_SRTP != 0)
-        srtp_active = pjsua_var.acc[call->acc_id].cfg.use_srtp;
-#else
-        srtp_active = PJ_FALSE;
-#endif
-
-        call->audio_idx = find_audio_index(rem_sdp, srtp_active);
-        if (call->audio_idx == -1) {
-            /* No audio in the offer. We can't accept this */
-            PJ_LOG(4,(THIS_FILE,
-                      "Unable to accept SDP offer without audio for call %d",
-                      call_id));
-            return PJMEDIA_SDP_EINMEDIA;
-        }
-    }
-
-    /* Media index must have been determined before */
-    pj_assert(call->audio_idx != -1);
-
-    /* Create media if it's not created. This could happen when call is
-     * currently on-hold
-     */
-    if (call->med_tp_st == PJSUA_MED_TP_IDLE) {
+    if (call->media[call->audio_idx].tp == NULL) {
         pjsip_role_e role;
         role = (rem_sdp ? PJSIP_ROLE_UAS : PJSIP_ROLE_UAC);
         status = pjsua_media_channel_init(call_id, role, call->secure_level, 
-                                          pool, rem_sdp, sip_status_code);
+                                          pool, rem_sdp, sip_err_code);
         if (status != PJ_SUCCESS)
             return status;
@@ -1340 +1616 @@
         sdp_neg_state = pjmedia_sdp_neg_get_state(call->inv->neg);
 
-    /* Get media socket info */
-    pjmedia_transport_info_init(&tpinfo);
-    pjmedia_transport_get_info(call->med_tp, &tpinfo);
-
-    /* Create SDP */
-    status = pjmedia_endpt_create_sdp(pjsua_var.med_endpt, pool, MAX_MEDIA,
-                                      &tpinfo.sock_info, &sdp);
-    if (status != PJ_SUCCESS) {
-        if (sip_status_code) *sip_status_code = 500;
+    /* Get one address to use in the origin field */
+    pj_bzero(&origin, sizeof(origin));
+    for (mi=0; mi<call->med_cnt; ++mi) {
+        pjmedia_transport_info tpinfo;
+
+        if (call->media[mi].tp == NULL)
+            continue;
+
+        pjmedia_transport_info_init(&tpinfo);
+        pjmedia_transport_get_info(call->media[mi].tp, &tpinfo);
+        pj_sockaddr_cp(&origin, &tpinfo.sock_info.rtp_addr_name);
+        break;
+    }
+
+    /* Create the base (blank) SDP */
+    status = pjmedia_endpt_create_base_sdp(pjsua_var.med_endpt, pool, NULL,
+                                           &origin, &sdp);
+    if (status != PJ_SUCCESS)
         return status;
-    }
-
-    /* If we're answering or updating the session with a new offer,
-     * and the selected media is not the first media
-     * in SDP, then fill in the unselected media with with zero port. 
-     * Otherwise we'll crash in transport_encode_sdp() because the media
-     * lines are not aligned between offer and answer.
-     */
-    if (call->audio_idx != 0 && 
-        (rem_sdp || sdp_neg_state==PJMEDIA_SDP_NEG_STATE_DONE))
-    {
-        unsigned i;
-        const pjmedia_sdp_session *ref_sdp = rem_sdp;
-
-        if (!ref_sdp) {
-            /* We are updating session with a new offer */
-            status = pjmedia_sdp_neg_get_active_local(call->inv->neg,
-                                                      &ref_sdp);
-            pj_assert(status == PJ_SUCCESS);
-        }
-
-        for (i=0; i<ref_sdp->media_count; ++i) {
-            const pjmedia_sdp_media *ref_m = ref_sdp->media[i];
-            pjmedia_sdp_media *m;
-
-            if ((int)i == call->audio_idx)
-                continue;
-
-            m = pjmedia_sdp_media_clone_deactivate(pool, ref_m);
-            if (i==sdp->media_count)
-                sdp->media[sdp->media_count++] = m;
-            else {
-                pj_array_insert(sdp->media, sizeof(sdp->media[0]),
-                                sdp->media_count, i, &m);
-                ++sdp->media_count;
+
+    /* Process each media line */
+    for (mi=0; mi<call->med_cnt; ++mi) {
+        pjsua_call_media *call_med = &call->media[mi];
+        pjmedia_sdp_media *m = NULL;
+        pjmedia_transport_info tpinfo;
+
+        if (call_med->tp == NULL) {
+            /*
+             * This media is deactivated. Just create a valid SDP with zero
+             * port.
+             */
+            m = PJ_POOL_ZALLOC_T(pool, pjmedia_sdp_media);
+            m->desc.transport = pj_str("RTP/AVP");
+            m->desc.fmt_count = 1;
+            m->conn = PJ_POOL_ZALLOC_T(pool, pjmedia_sdp_conn);
+            m->conn->net_type = pj_str("IN");
+            m->conn->addr_type = pj_str("IP4");
+            m->conn->addr = pj_str("127.0.0.1");
+
+            switch (call_med->type) {
+            case PJMEDIA_TYPE_AUDIO:
+                m->desc.media = pj_str("audio");
+                m->desc.fmt[0] = pj_str("0");
+                break;
+            case PJMEDIA_TYPE_VIDEO:
+                m->desc.media = pj_str("video");
+                m->desc.fmt[0] = pj_str("31");
+                break;
+            default:
+                if (rem_sdp && mi < rem_sdp->media_count) {
+                    pj_strdup(pool, &m->desc.media,
+                              &rem_sdp->media[mi]->desc.media);
+                    pj_strdup(pool, &m->desc.fmt[0],
+                              &rem_sdp->media[mi]->desc.fmt[0]);
+                } else {
+                    pj_assert(!"Invalid call_med media type");
+                    return PJ_EBUG;
+                }
             }
+
+            sdp->media[sdp->media_count++] = m;
+            continue;
+        }
+
+        /* Get transport address info */
+        pjmedia_transport_info_init(&tpinfo);
+        pjmedia_transport_get_info(call_med->tp, &tpinfo);
+
+        /* Ask pjmedia endpoint to create SDP media line */
+        switch (call_med->type) {
+        case PJMEDIA_TYPE_AUDIO:
+            status = pjmedia_endpt_create_audio_sdp(pjsua_var.med_endpt, pool,
+                                                    &tpinfo.sock_info, 0, &m);
+            break;
+        case PJMEDIA_TYPE_VIDEO:
+            status = pjmedia_endpt_create_video_sdp(pjsua_var.med_endpt, pool,
+                                                    &tpinfo.sock_info, 0, &m);
+            break;
+        default:
+            pj_assert(!"Invalid call_med media type");
+            return PJ_EBUG;
+        }
+
+        if (status != PJ_SUCCESS)
+            return status;
+
+        sdp->media[sdp->media_count++] = m;
+
+        /* Give to transport */
+        status = pjmedia_transport_encode_sdp(call_med->tp, pool,
+                                              sdp, rem_sdp, mi);
+        if (status != PJ_SUCCESS) {
+            if (sip_err_code) *sip_err_code = PJSIP_SC_NOT_ACCEPTABLE;
+            return status;
         }
     }
@@ -1413 +1737 @@
     }
 
-    /* Give the SDP to media transport */
-    status = pjmedia_transport_encode_sdp(call->med_tp, pool, sdp, rem_sdp, 
-                                          call->audio_idx);
-    if (status != PJ_SUCCESS) {
-        if (sip_status_code) *sip_status_code = PJSIP_SC_NOT_ACCEPTABLE;
-        return status;
-    }
-
-#if defined(PJMEDIA_HAS_SRTP) && (PJMEDIA_HAS_SRTP != 0)
+
+#if DISABLED_FOR_TICKET_1185 && defined(PJMEDIA_HAS_SRTP) && (PJMEDIA_HAS_SRTP != 0)
     /* Check if SRTP is in optional mode and configured to use duplicated
      * media, i.e: secured and unsecured version, in the SDP offer.
@@ -1434 +1751 @@
             pjmedia_sdp_media *m = sdp->media[i];
 
-            /* Check if this media is unsecured but has SDP "crypto" 
+            /* Check if this media is unsecured but has SDP "crypto"
              * attribute.
              */
@@ -1440 +1757 @@
                 pjmedia_sdp_media_find_attr2(m, "crypto", NULL) != NULL)
             {
-                if (i == (unsigned)call->audio_idx && 
+                if (i == (unsigned)call->audio_idx &&
                     sdp_neg_state == PJMEDIA_SDP_NEG_STATE_DONE)
                 {
@@ -1463 +1780 @@
                     /* Insert the new media before the unsecured media */
                     if (sdp->media_count < PJMEDIA_MAX_SDP_MEDIA) {
-                        pj_array_insert(sdp->media, sizeof(new_m), 
+                        pj_array_insert(sdp->media, sizeof(new_m),
                                         sdp->media_count, i, &new_m);
                         ++sdp->media_count;
@@ -1474 +1791 @@
 #endif
 
-    /* Update currently advertised RTP source address */
-    pj_memcpy(&call->med_rtp_addr, &tpinfo.sock_info.rtp_addr_name, 
-              sizeof(pj_sockaddr));
-
     *p_sdp = sdp;
     return PJ_SUCCESS;
@@ -1486 +1799 @@
 {
     pjsua_call *call = &pjsua_var.calls[call_id];
-
-    if (call->conf_slot != PJSUA_INVALID_ID) {
-        if (pjsua_var.mconf) {
-            pjsua_conf_remove_port(call->conf_slot);
-        }
-        call->conf_slot = PJSUA_INVALID_ID;
-    }
-
-    if (call->session) {
-        pjmedia_rtcp_stat stat;
-
-        if ((call->media_dir & PJMEDIA_DIR_ENCODING) &&
-            (pjmedia_session_get_stream_stat(call->session, 0, &stat) 
-             == PJ_SUCCESS))
+    unsigned mi;
+
+    for (mi=0; mi<call->med_cnt; ++mi) {
+        pjsua_call_media *call_med = &call->media[mi];
+
+        if (call_med->type == PJMEDIA_TYPE_AUDIO) {
+            pjmedia_stream *strm = call_med->strm.a.stream;
+            pjmedia_rtcp_stat stat;
+
+            if (strm) {
+                if (call_med->strm.a.conf_slot != PJSUA_INVALID_ID) {
+                    if (pjsua_var.mconf) {
+                        pjsua_conf_remove_port(call_med->strm.a.conf_slot);
+                    }
+                    call_med->strm.a.conf_slot = PJSUA_INVALID_ID;
+                }
+
+                if ((call_med->dir & PJMEDIA_DIR_ENCODING) &&
+                    (pjmedia_stream_get_stat(strm, &stat) == PJ_SUCCESS))
+                {
+                    /* Save RTP timestamp & sequence, so when media session is
+                     * restarted, those values will be restored as the initial
+                     * RTP timestamp & sequence of the new media session. So in
+                     * the same call session, RTP timestamp and sequence are
+                     * guaranteed to be contigue.
+                     */
+                    call_med->rtp_tx_seq_ts_set = 1 | (1 << 1);
+                    call_med->rtp_tx_seq = stat.rtp_tx_last_seq;
+                    call_med->rtp_tx_ts = stat.rtp_tx_last_ts;
+                }
+
+                if (pjsua_var.ua_cfg.cb.on_stream_destroyed) {
+                    pjsua_var.ua_cfg.cb.on_stream_destroyed(call_id, strm, mi);
+                }
+
+                pjmedia_stream_destroy(strm);
+                call_med->strm.a.stream = NULL;
+
+                PJ_LOG(4,(THIS_FILE, "Media session call%02d:%d is destroyed",
+                                     call_id, mi));
+            }
+        }
+        call_med->state = PJSUA_CALL_MEDIA_NONE;
+    }
+}
+
+pj_status_t pjsua_media_channel_deinit(pjsua_call_id call_id)
+{
+    pjsua_call *call = &pjsua_var.calls[call_id];
+    unsigned mi;
+
+    stop_media_session(call_id);
+
+    for (mi=0; mi<call->med_cnt; ++mi) {
+        pjsua_call_media *call_med = &call->media[mi];
+
+        if (call_med->tp_st != PJSUA_MED_TP_IDLE) {
+            pjmedia_transport_media_stop(call_med->tp);
+            call_med->tp_st = PJSUA_MED_TP_IDLE;
+        }
+
+        if (call_med->tp_orig && call_med->tp &&
+                call_med->tp != call_med->tp_orig)
         {
-            /* Save RTP timestamp & sequence, so when media session is 
-             * restarted, those values will be restored as the initial 
-             * RTP timestamp & sequence of the new media session. So in 
-             * the same call session, RTP timestamp and sequence are 
-             * guaranteed to be contigue.
-             */
-            call->rtp_tx_seq_ts_set = 1 | (1 << 1);
-            call->rtp_tx_seq = stat.rtp_tx_last_seq;
-            call->rtp_tx_ts = stat.rtp_tx_last_ts;
-        }
-
-        if (pjsua_var.ua_cfg.cb.on_stream_destroyed) {
-            pjsua_var.ua_cfg.cb.on_stream_destroyed(call_id, call->session, 0);
-        }
-
-        pjmedia_session_destroy(call->session);
-        call->session = NULL;
-
-        PJ_LOG(4,(THIS_FILE, "Media session for call %d is destroyed", 
-                             call_id));
-
-    }
-
-    call->media_st = PJSUA_CALL_MEDIA_NONE;
-}
-
-pj_status_t pjsua_media_channel_deinit(pjsua_call_id call_id)
-{
-    pjsua_call *call = &pjsua_var.calls[call_id];
-
-    stop_media_session(call_id);
-
-    if (call->med_tp_st != PJSUA_MED_TP_IDLE) {
-        pjmedia_transport_media_stop(call->med_tp);
-        call->med_tp_st = PJSUA_MED_TP_IDLE;
-    }
-
-    if (call->med_orig && call->med_tp && call->med_tp != call->med_orig) {
-        pjmedia_transport_close(call->med_tp);
-        call->med_tp = call->med_orig;
+            pjmedia_transport_close(call_med->tp);
+            call_med->tp = call_med->tp_orig;
+        }
     }
 
@@ -1570 +1895 @@
 
 
-pj_status_t pjsua_media_channel_update(pjsua_call_id call_id,
-                                       const pjmedia_sdp_session *local_sdp,
-                                       const pjmedia_sdp_session *remote_sdp)
-{
-    int prev_media_st = 0;
-    pjsua_call *call = &pjsua_var.calls[call_id];
-    pjmedia_session_info sess_info;
-    pjmedia_stream_info *si = NULL;
+static pj_status_t audio_channel_update(pjsua_call_media *call_med,
+                                        pj_pool_t *tmp_pool,
+                                        const pjmedia_sdp_session *local_sdp,
+                                        const pjmedia_sdp_session *remote_sdp)
+{
+    pjsua_call *call = call_med->call;
+    pjmedia_stream_info the_si, *si = &the_si;
     pjmedia_port *media_port;
+    unsigned strm_idx = call_med->idx;
     pj_status_t status;
-
-    if (!pjsua_var.med_endpt) {
-        /* We're being shutdown */
-        return PJ_EBUSY;
-    }
-
-    /* Destroy existing media session, if any. */
-    prev_media_st = call->media_st;
-    stop_media_session(call->index);
-
-    /* Create media session info based on SDP parameters. 
-     */    
-    status = pjmedia_session_info_from_sdp( call->inv->pool_prov, 
-                                            pjsua_var.med_endpt, 
-                                            PJMEDIA_MAX_SDP_MEDIA, &sess_info,
-                                            local_sdp, remote_sdp);
+    
+    status = pjmedia_stream_info_from_sdp(si, tmp_pool, pjsua_var.med_endpt,
+                                          local_sdp, remote_sdp, strm_idx);
     if (status != PJ_SUCCESS)
         return status;
 
-    /* Update audio index from the negotiated SDP */
-    call->audio_idx = find_audio_index(local_sdp, PJ_TRUE);
-
-    /* Find which session is audio */
-    PJ_ASSERT_RETURN(call->audio_idx != -1, PJ_EBUG);
-    PJ_ASSERT_RETURN(call->audio_idx < (int)sess_info.stream_cnt, PJ_EBUG);
-    si = &sess_info.stream_info[call->audio_idx];
-    
-    /* Reset session info with only one media stream */
-    sess_info.stream_cnt = 1;
-    if (si != &sess_info.stream_info[0]) {
-        pj_memcpy(&sess_info.stream_info[0], si, sizeof(pjmedia_stream_info));
-        si = &sess_info.stream_info[0];
-    }
-
     /* Check if no media is active */
-    if (sess_info.stream_cnt == 0 || si->dir == PJMEDIA_DIR_NONE)
-    {
+    if (si->dir == PJMEDIA_DIR_NONE) {
         /* Call media state */
-        call->media_st = PJSUA_CALL_MEDIA_NONE;
+        call_med->state = PJSUA_CALL_MEDIA_NONE;
 
         /* Call media direction */
-        call->media_dir = PJMEDIA_DIR_NONE;
-
-        /* Don't stop transport because we need to transmit keep-alives, and
-         * also to prevent restarting ICE negotiation. See
-         *  http://trac.pjsip.org/repos/ticket/1094
-         */
-#if 0
-        /* Shutdown transport's session */
-        pjmedia_transport_media_stop(call->med_tp);
-        call->med_tp_st = PJSUA_MED_TP_IDLE;
-
-        /* No need because we need keepalive? */
-
-        /* Close upper entry of transport stack */
-        if (call->med_orig && (call->med_tp != call->med_orig)) {
-            pjmedia_transport_close(call->med_tp);
-            call->med_tp = call->med_orig;
-        }
-#endif
+        call_med->dir = PJMEDIA_DIR_NONE;
 
     } else {
@@ -1645 +1923 @@
 
         /* Start/restart media transport */
-        status = pjmedia_transport_media_start(call->med_tp, 
-                                               call->inv->pool_prov,
-                                               local_sdp, remote_sdp,
-                                               call->audio_idx);
+        status = pjmedia_transport_media_start(call_med->tp,
+                                               tmp_pool, local_sdp,
+                                               remote_sdp, strm_idx);
         if (status != PJ_SUCCESS)
             return status;
 
-        call->med_tp_st = PJSUA_MED_TP_RUNNING;
+        call_med->tp_st = PJSUA_MED_TP_RUNNING;
 
         /* Get remote SRTP usage policy */
         pjmedia_transport_info_init(&tp_info);
-        pjmedia_transport_get_info(call->med_tp, &tp_info);
+        pjmedia_transport_get_info(call_med->tp, &tp_info);
         if (tp_info.specific_info_cnt > 0) {
             unsigned i;
@@ -1665 +1942 @@
                                 (pjmedia_srtp_info*) tp_info.spc_info[i].buffer;
 
-                    call->rem_srtp_use = srtp_info->peer_use;
+                    call_med->rem_srtp_use = srtp_info->peer_use;
                     break;
                 }
@@ -1694 +1971 @@
 
         /* Set SSRC */
-        si->ssrc = call->ssrc;
+        si->ssrc = call_med->ssrc;
 
         /* Set RTP timestamp & sequence, normally these value are intialized
@@ -1701 +1978 @@
          * contigue.
          */
-        si->rtp_ts = call->rtp_tx_ts;
-        si->rtp_seq = call->rtp_tx_seq;
-        si->rtp_seq_ts_set = call->rtp_tx_seq_ts_set;
+        si->rtp_ts = call_med->rtp_tx_ts;
+        si->rtp_seq = call_med->rtp_tx_seq;
+        si->rtp_seq_ts_set = call_med->rtp_tx_seq_ts_set;
 
 #if defined(PJMEDIA_STREAM_ENABLE_KA) && PJMEDIA_STREAM_ENABLE_KA!=0
@@ -1711 +1988 @@
 
         /* Create session based on session info. */
-        status = pjmedia_session_create( pjsua_var.med_endpt, &sess_info,
-                                         &call->med_tp,
-                                         call, &call->session );
+        status = pjmedia_stream_create(pjsua_var.med_endpt, NULL, si,
+                                       call_med->tp, NULL,
+                                       &call_med->strm.a.stream);
+        if (status != PJ_SUCCESS) {
+            return status;
+        }
+
+        /* Start stream */
+        status = pjmedia_stream_start(call_med->strm.a.stream);
         if (status != PJ_SUCCESS) {
             return status;
@@ -1722 +2005 @@
          */
         if (pjsua_var.ua_cfg.cb.on_dtmf_digit) {
-            pjmedia_session_set_dtmf_callback(call->session, 0, 
-                                              &dtmf_callback, 
-                                              (void*)(long)(call->index));
+            pjmedia_stream_set_dtmf_callback(call_med->strm.a.stream,
+                                             &dtmf_callback,
+                                             (void*)(long)(call->index));
         }
 
@@ -1730 +2013 @@
          * We need the port interface to add to the conference bridge.
          */
-        pjmedia_session_get_port(call->session, 0, &media_port);
+        pjmedia_stream_get_port(call_med->strm.a.stream, &media_port);
 
         /* Notify application about stream creation.
@@ -1737 +2020 @@
          */
         if (pjsua_var.ua_cfg.cb.on_stream_created) {
-            pjsua_var.ua_cfg.cb.on_stream_created(call_id, call->session,
-                                                  0, &media_port);
+            pjsua_var.ua_cfg.cb.on_stream_created(call->index,
+                                                  call_med->strm.a.stream,
+                                                  strm_idx, &media_port);
         }
 
@@ -1759 +2043 @@
                                             media_port, 
                                             &port_name,
-                                            (unsigned*)&call->conf_slot);
+                                            (unsigned*)
+                                            &call_med->strm.a.conf_slot);
             if (status != PJ_SUCCESS) {
                 return status;
@@ -1766 +2051 @@
 
         /* Call media direction */
-        call->media_dir = si->dir;
+        call_med->dir = si->dir;
 
         /* Call media state */
         if (call->local_hold)
-            call->media_st = PJSUA_CALL_MEDIA_LOCAL_HOLD;
-        else if (call->media_dir == PJMEDIA_DIR_DECODING)
-            call->media_st = PJSUA_CALL_MEDIA_REMOTE_HOLD;
+            call_med->state = PJSUA_CALL_MEDIA_LOCAL_HOLD;
+        else if (call_med->dir == PJMEDIA_DIR_DECODING)
+            call_med->state = PJSUA_CALL_MEDIA_REMOTE_HOLD;
         else
-            call->media_st = PJSUA_CALL_MEDIA_ACTIVE;
+            call_med->state = PJSUA_CALL_MEDIA_ACTIVE;
     }
 
@@ -1781 +2066 @@
         char info[80];
         int info_len = 0;
-        unsigned i;
-
-        for (i=0; i<sess_info.stream_cnt; ++i) {
-            int len;
-            const char *dir;
-            pjmedia_stream_info *strm_info = &sess_info.stream_info[i];
-
-            switch (strm_info->dir) {
-            case PJMEDIA_DIR_NONE:
-                dir = "inactive";
-                break;
-            case PJMEDIA_DIR_ENCODING:
-                dir = "sendonly";
-                break;
-            case PJMEDIA_DIR_DECODING:
-                dir = "recvonly";
-                break;
-            case PJMEDIA_DIR_ENCODING_DECODING:
-                dir = "sendrecv";
-                break;
-            default:
-                dir = "unknown";
-                break;
+        int len;
+        const char *dir;
+
+        switch (si->dir) {
+        case PJMEDIA_DIR_NONE:
+            dir = "inactive";
+            break;
+        case PJMEDIA_DIR_ENCODING:
+            dir = "sendonly";
+            break;
+        case PJMEDIA_DIR_DECODING:
+            dir = "recvonly";
+            break;
+        case PJMEDIA_DIR_ENCODING_DECODING:
+            dir = "sendrecv";
+            break;
+        default:
+            dir = "unknown";
+            break;
+        }
+        len = pj_ansi_sprintf( info+info_len,
+                               ", stream #%d: %.*s (%s)", strm_idx,
+                               (int)si->fmt.encoding_name.slen,
+                               si->fmt.encoding_name.ptr,
+                               dir);
+        if (len > 0)
+            info_len += len;
+        PJ_LOG(4,(THIS_FILE,"Media updates%s", info));
+    }
+
+    return PJ_SUCCESS;
+}
+
+pj_status_t pjsua_media_channel_update(pjsua_call_id call_id,
+                                       const pjmedia_sdp_session *local_sdp,
+                                       const pjmedia_sdp_session *remote_sdp)
+{
+    pjsua_call *call = &pjsua_var.calls[call_id];
+    pj_pool_t *tmp_pool = call->inv->pool_prov;
+    unsigned mi;
+    pj_status_t status;
+
+    if (pjsua_get_state() != PJSUA_STATE_RUNNING)
+        return PJ_EBUSY;
+
+    /* Destroy existing media session, if any. */
+    stop_media_session(call->index);
+
+    /* Reset audio_idx first */
+    call->audio_idx = -1;
+
+    /* Process each media stream */
+    for (mi=0; mi < call->med_cnt; ++mi) {
+        pjsua_call_media *call_med = &call->media[mi];
+
+        if (mi > local_sdp->media_count ||
+            mi > remote_sdp->media_count)
+        {
+            /* Something is wrong */
+            PJ_LOG(1,(THIS_FILE, "Error updating media for call %d: "
+                      "invalid media index %d in SDP", call_id, mi));
+            return PJMEDIA_SDP_EINSDP;
+        }
+
+        switch (call_med->type) {
+        case PJMEDIA_TYPE_AUDIO:
+            status = audio_channel_update(call_med, tmp_pool,
+                                          local_sdp, remote_sdp);
+            if (call->audio_idx==-1 && status==PJ_SUCCESS &&
+                    call_med->strm.a.stream)
+            {
+                call->audio_idx = mi;
             }
-            len = pj_ansi_sprintf( info+info_len,
-                                   ", stream #%d: %.*s (%s)", i,
-                                   (int)strm_info->fmt.encoding_name.slen,
-                                   strm_info->fmt.encoding_name.ptr,
-                                   dir);
-            if (len > 0)
-                info_len += len;
-        }
-        PJ_LOG(4,(THIS_FILE,"Media updates%s", info));
+            break;
+        case PJMEDIA_TYPE_VIDEO:
+            PJ_LOG(4,(THIS_FILE, "-x-x-x-x-  Updating video for stream %d", mi));
+            break;
+        default:
+            break;
+        }
+
+        if (status != PJ_SUCCESS) {
+            PJ_PERROR(1,(THIS_FILE, status, "Error updating media call%02:%d",
+                         call_id, mi));
+        }
     }