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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
This commit is contained in:
Olle Johansson 2008-02-18 17:12:13 +00:00
parent e9d0cda77d
commit f04d5fb83d
2 changed files with 35 additions and 12 deletions
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22
channels/chan_sip.c
View File

@ -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);

25
main/channel.c
View File

@ -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;