Changeset 1641 for pjproject/trunk/pjsip-apps/src/symbian_ua/ua.cpp

Timestamp:

Dec 28, 2007 6:55:02 PM (16 years ago)

Author:

bennylp

Message:

Ticket #435: Fixed and tested audio on Symbian device

File:

• pjproject/trunk/pjsip-apps/src/symbian_ua/ua.cpp (modified) (6 diffs)

pjproject/trunk/pjsip-apps/src/symbian_ua/ua.cpp

@@ -32 +32 @@
 // Destination URI (to make call, or to subscribe presence)
 //
-#define SIP_DST_URI     "sip:192.168.0.7:5061"
+#define SIP_DST_URI     "sip:192.168.0.11:5060"
 
 //
@@ -58 +58 @@
 //
 // STUN server
-#if 1
+#if 0
         // Use this to have the STUN server resolved normally
 #   define STUN_DOMAIN  NULL
@@ -75 +75 @@
 // Use ICE?
 //
-#define USE_ICE         1
+#define USE_ICE         0
 
 
@@ -321 +321 @@
     med_cfg.has_ioqueue = PJ_FALSE;
     med_cfg.clock_rate = 8000;
+    med_cfg.audio_frame_ptime = 40;
     med_cfg.ec_tail_len = 0;
     med_cfg.enable_ice = USE_ICE;
@@ -453 +454 @@
             "  D    Dump states detail\n"
             "  P    Dump pool factory\n"
+            "  l    Start loopback audio device\n"
+            "  L    Stop loopback audio device\n"
             "  m    Call " SIP_DST_URI "\n"
             "  a    Answer call\n"
@@ -482 +485 @@
             pj_pool_factory_dump(pjsua_get_pool_factory(), PJ_TRUE);
             break;
+    case 'l':
+                pjsua_conf_connect(0, 0);
+            break;
+    case 'L':
+                pjsua_conf_disconnect(0, 0);
+            break;
     case 'm':
             if (g_call_id != PJSUA_INVALID_ID) {