[asterisk-commits] oej: trunk r103772 - in /trunk: channels/chan_sip.c main/channel.c

[SVN commits to the Asterisk project]

Author: oej
Date: Mon Feb 18 11:12:13 2008
New Revision: 103772

URL: http://svn.digium.com/view/asterisk?view=rev&rev=103772
Log:
Make sure we can set up calls without audio (text+video).
And ... it works!


Modified:
    trunk/channels/chan_sip.c
    trunk/main/channel.c

Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=103772&r1=103771&r2=103772
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Mon Feb 18 11:12:13 2008
@@ -7804,6 +7804,7 @@
 
 	int x;
 	int capability;
+	int needaudio = FALSE;
 	int needvideo = FALSE;
 	int needtext = FALSE;
 	int debug = sip_debug_test_pvt(p);
@@ -7846,6 +7847,10 @@
 		ast_str_append(&a_audio, 0, "a=rtpmap:%d %s/%d\r\n", 191, "t38", 8000);
 	}
 #endif
+
+	/* Check if we need audio */
+	if (capability & AST_FORMAT_AUDIO_MASK)
+		needaudio = TRUE;
 
 	/* Check if we need video in this call */
 	if ((capability & AST_FORMAT_VIDEO_MASK) && !p->novideo) {
@@ -7941,7 +7946,7 @@
 		alreadysent |= codec;
 	}
 
-	/* Start by sending our preferred audio codecs */
+	/* Start by sending our preferred audio/video codecs */
 	for (x = 0; x < 32; x++) {
 		int codec;
 
@@ -8008,14 +8013,17 @@
 			a_audio->len - a_audio->used < 2 || a_video->len - a_video->used < 2)
 		ast_log(LOG_WARNING, "SIP SDP may be truncated due to undersized buffer!!\n");
 
- 	ast_str_append(&m_audio, 0, "\r\n");
+	if (needaudio)
+ 		ast_str_append(&m_audio, 0, "\r\n");
  	if (needvideo)
  		ast_str_append(&m_video, 0, "\r\n");
  	if (needtext)
  		ast_str_append(&m_text, 0, "\r\n");
 
  	len = strlen(version) + strlen(subject) + strlen(owner) +
-		strlen(connection) + strlen(stime) + m_audio->used + a_audio->used + strlen(hold);
+		strlen(connection) + strlen(stime);
+	if (needaudio)
+		len += m_audio->used + a_audio->used + strlen(hold);
  	if (needvideo) /* only if video response is appropriate */
  		len += m_video->used + a_video->used + strlen(bandwidth) + strlen(hold);
  	if (needtext) /* only if text response is appropriate */
@@ -8030,9 +8038,11 @@
 	if (needvideo)	 	/* only if video response is appropriate */
 		add_line(resp, bandwidth);
 	add_line(resp, stime);
-	add_line(resp, m_audio->str);
-	add_line(resp, a_audio->str);
-	add_line(resp, hold);
+	if (needaudio) {
+		add_line(resp, m_audio->str);
+		add_line(resp, a_audio->str);
+		add_line(resp, hold);
+	}
 	if (needvideo) { /* only if video response is appropriate */
 		add_line(resp, m_video->str);
 		add_line(resp, a_video->str);

Modified: trunk/main/channel.c
URL: http://svn.digium.com/view/asterisk/trunk/main/channel.c?view=diff&rev=103772&r1=103771&r2=103772
==============================================================================
--- trunk/main/channel.c (original)
+++ trunk/main/channel.c Mon Feb 18 11:12:13 2008
@@ -3094,6 +3094,9 @@
 {
 	int native;
 	int res;
+
+	if (!fmt || !native)	/* No audio requested */
+		return 0;	/* Let's try a call without any sounds (video, text) */
 	
 	/* Make sure we only consider audio */
 	fmt &= AST_FORMAT_AUDIO_MASK;
@@ -3337,12 +3340,17 @@
 
 		capabilities = chan->tech->capabilities;
 		fmt = format & AST_FORMAT_AUDIO_MASK;
-		res = ast_translator_best_choice(&fmt, &capabilities);
-		if (res < 0) {
-			ast_log(LOG_WARNING, "No translator path exists for channel type %s (native 0x%x) to 0x%x\n", type, chan->tech->capabilities, format);
-			*cause = AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
-			AST_RWLIST_UNLOCK(&channels);
-			return NULL;
+		if (fmt) {
+			/* We have audio - is it possible to connect the various calls to each other? 
+				(Avoid this check for calls without audio, like text+video calls)
+			*/
+			res = ast_translator_best_choice(&fmt, &capabilities);
+			if (res < 0) {
+				ast_log(LOG_WARNING, "No translator path exists for channel type %s (native 0x%x) to 0x%x\n", type, chan->tech->capabilities, format);
+				*cause = AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
+				AST_RWLIST_UNLOCK(&channels);
+				return NULL;
+			}
 		}
 		AST_RWLIST_UNLOCK(&channels);
 		if (!chan->tech->requester)
@@ -3483,6 +3491,11 @@
 	/* Set up translation from the 'from' channel to the 'to' channel */
 	src = from->nativeformats;
 	dst = to->nativeformats;
+
+	/* If there's no audio in this call, don't bother with trying to find a translation path */
+	if ((src & AST_FORMAT_AUDIO_MASK) == 0 || (dst & AST_FORMAT_AUDIO_MASK) == 0)
+		return 0;
+
 	if (ast_translator_best_choice(&dst, &src) < 0) {
 		ast_log(LOG_WARNING, "No path to translate from %s(%d) to %s(%d)\n", from->name, src, to->name, dst);
 		return -1;