Make sure we can set up calls without audio (text+video).

And ... it works!

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103772 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 9d755c6..6f4ed04 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -7804,6 +7804,7 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
 
        int x;
        int capability;
+       int needaudio = FALSE;
        int needvideo = FALSE;
        int needtext = FALSE;
        int debug = sip_debug_test_pvt(p);
@@ -7847,6 +7848,10 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
        }
 #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) {
                if (p->vrtp) {
@@ -7941,7 +7946,7 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
                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 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
                        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 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
        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);

diff --git a/main/channel.c b/main/channel.c
index f8f1fe6..bbfcfdd 100644 (file)
--- a/main/channel.c
+++ b/main/channel.c
@@ -3094,6 +3094,9 @@ static int set_format(struct ast_channel *chan, int fmt, int *rawformat, int *fo
 {
        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 @@ struct ast_channel *ast_request(const char *type, int format, void *data, int *c
 
                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 @@ static int ast_channel_make_compatible_helper(struct ast_channel *from, struct a
        /* 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;