Changeset 2079

Timestamp:

Jun 27, 2008 9:19:44 PM (16 years ago)

Author:

bennylp

Message:

Fix handling of multiple media lines in the incoming SDP offer. Now pjsua-lib will be able to select the best media line to handle

Location:

pjproject/trunk/pjsip

Files:

• include/pjsua-lib/pjsua_internal.h (modified) (1 diff)
• src/pjsua-lib/pjsua_call.c (modified) (1 diff)
• src/pjsua-lib/pjsua_media.c (modified) (9 diffs)

pjproject/trunk/pjsip/include/pjsua-lib/pjsua_internal.h

@@ -64 +64 @@
     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                           */
     int                  conf_slot; /**< Slot # in conference bridge.       */

pjproject/trunk/pjsip/src/pjsua-lib/pjsua_call.c

@@ -105 +105 @@
     call->user_data = NULL;
     call->session = NULL;
+    call->audio_idx = -1;
     call->ssrc = pj_rand();
     call->xfer_sub = NULL;

pjproject/trunk/pjsip/src/pjsua-lib/pjsua_media.c

@@ -852 +852 @@
                                      int *sip_err_code)
 {
-    enum { MEDIA_IDX = 0 };
     pjsua_call *call = &pjsua_var.calls[call_id];
     pj_status_t status;
@@ -859 +858 @@
     pjsua_acc *acc = &pjsua_var.acc[call->acc_id];
     pjmedia_srtp_setting srtp_opt;
-    pjmedia_transport *srtp;
+    pjmedia_transport *srtp = NULL;
 #endif
 
@@ -915 +914 @@
 #endif
 
+    /* Find out which media line in SDP that we support. If we are offerer,
+     * audio will be at index 0 in SDP. 
+     */
+    if (rem_sdp == 0) {
+        call->audio_idx = 0;
+    } 
+    /* Otherwise find out the candidate audio media line in SDP */
+    else {
+        unsigned i;
+        pj_bool_t srtp_active;
+
+#if defined(PJMEDIA_HAS_SRTP) && (PJMEDIA_HAS_SRTP != 0)
+        srtp_active = acc->cfg.use_srtp && srtp != NULL;
+#else
+        srtp_active = PJ_FALSE;
+#endif
+
+        /* Media count must have been checked */
+        pj_assert(rem_sdp->media_count != 0);
+
+        for (i=0; i<rem_sdp->media_count; ++i) {
+            const pjmedia_sdp_media *m = rem_sdp->media[i];
+
+            /* Skip if media is not audio */
+            if (pj_stricmp2(&m->desc.media, "audio") != 0)
+                continue;
+
+            /* Skip if media is 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 (call->audio_idx == -1) {
+                call->audio_idx = i;
+            } else {
+                /* We've found multiple candidates. This could happen
+                 * e.g. when remote is offering both RTP/AVP and RTP/AVP,
+                 * or when remote for some reason offers two audio.
+                 */
+
+                if (srtp_active &&
+                    pj_stricmp2(&m->desc.transport, "RTP/SAVP")==0)
+                {
+                    /* Prefer RTP/SAVP when our media transport is SRTP */
+                    call->audio_idx = i;
+                } else if (!srtp_active &&
+                           pj_stricmp2(&m->desc.transport, "RTP/AVP")==0)
+                {
+                    /* Prefer RTP/AVP when our media transport is NOT SRTP */
+                    call->audio_idx = i;
+                }
+            }
+        }
+    }
+
+    /* Reject offer if we couldn't find a good m=audio line in offer */
+    if (call->audio_idx < 0) {
+        if (sip_err_code) *sip_err_code = PJSIP_SC_NOT_ACCEPTABLE;
+        pjsua_media_channel_deinit(call_id);
+        return PJSIP_ERRNO_FROM_SIP_STATUS(PJSIP_SC_NOT_ACCEPTABLE);
+    }
+
+    PJ_LOG(4,(THIS_FILE, "Media index %d selected for call %d",
+              call->audio_idx, call->index));
+
     /* Create the media transport */
     status = pjmedia_transport_media_create(call->med_tp, tmp_pool, 0,
-                                            rem_sdp, MEDIA_IDX);
+                                            rem_sdp, call->audio_idx);
     if (status != PJ_SUCCESS) {
         if (sip_err_code) *sip_err_code = PJSIP_SC_NOT_ACCEPTABLE;
@@ -934 +1004 @@
                                            int *sip_status_code)
 {
-    enum { MAX_MEDIA = 1, MEDIA_IDX = 0 };
+    enum { MAX_MEDIA = 1 };
     pjmedia_sdp_session *sdp;
     pjmedia_transport_info tpinfo;
@@ -946 +1016 @@
         return PJ_EBUSY;
     }
+
+    /* 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
@@ -971 +1044 @@
     }
 
+    /* If we're answering 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 (rem_sdp && call->audio_idx != 0) {
+        unsigned i;
+
+        for (i=0; i<rem_sdp->media_count; ++i) {
+            const pjmedia_sdp_media *rem_m = rem_sdp->media[i];
+            pjmedia_sdp_media *m;
+            const pjmedia_sdp_attr *a;
+
+            if ((int)i == call->audio_idx)
+                continue;
+
+            m = PJ_POOL_ZALLOC_T(pool, pjmedia_sdp_media);
+            pj_strdup(pool, &m->desc.media, &rem_m->desc.media);
+            pj_strdup(pool, &m->desc.transport, &rem_m->desc.transport);
+            m->desc.port = 0;
+
+            /* Add one format, copy from the offer. And copy the corresponding
+             * rtpmap and fmtp attributes too.
+             */
+            m->desc.fmt_count = 1;
+            pj_strdup(pool, &m->desc.fmt[0], &rem_m->desc.fmt[0]);
+            if ((a=pjmedia_sdp_attr_find2(rem_m->attr_count, rem_m->attr,
+                                          "rtpmap", &m->desc.fmt[0])) != NULL)
+            {
+                m->attr[m->attr_count++] = pjmedia_sdp_attr_clone(pool, a);
+            }
+            if ((a=pjmedia_sdp_attr_find2(rem_m->attr_count, rem_m->attr,
+                                          "fmtp", &m->desc.fmt[0])) != NULL)
+            {
+                m->attr[m->attr_count++] = pjmedia_sdp_attr_clone(pool, a);
+            }
+
+            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;
+            }
+        }
+    }
+
     /* Add NAT info in the SDP */
     if (pjsua_var.ua_cfg.nat_type_in_sdp) {
@@ -997 +1117 @@
     /* Give the SDP to media transport */
     status = pjmedia_transport_encode_sdp(call->med_tp, pool, sdp, rem_sdp, 
-                                          MEDIA_IDX);
+                                          call->audio_idx);
     if (status != PJ_SUCCESS) {
         if (sip_status_code) *sip_status_code = PJSIP_SC_NOT_ACCEPTABLE;
@@ -1079 +1199 @@
                                        const pjmedia_sdp_session *remote_sdp)
 {
-    unsigned i;
     int prev_media_st = 0;
     pjsua_call *call = &pjsua_var.calls[call_id];
@@ -1100 +1219 @@
         return status;
 
-    /* Find which session is audio (we only support audio for now) */
-    for (i=0; i < sess_info.stream_cnt; ++i) {
-        if (sess_info.stream_info[i].type == PJMEDIA_TYPE_AUDIO &&
-            (sess_info.stream_info[i].proto == PJMEDIA_TP_PROTO_RTP_AVP ||
-             sess_info.stream_info[i].proto == PJMEDIA_TP_PROTO_RTP_SAVP))
-        {
-            si = &sess_info.stream_info[i];
-            break;
-        }
-    }
-
-    if (si == NULL) {
-        /* Not found */
-        return PJMEDIA_EINVALIMEDIATYPE;
-    }
-
+    /* 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 */