diff options
| -rw-r--r-- | channels/chan_sip.c | 22 | ||||
| -rw-r--r-- | main/channel.c | 25 |
2 files changed, 35 insertions, 12 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 9d755c684..6f4ed046b 100644 --- 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 f8f1fe637..bbfcfdd4a 100644 --- 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; |