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Merged revisions 314017 via svnmerge from

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https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r314017 | dvossel | 2011-04-18 08:41:06 -0500 (Mon, 18 Apr 2011) | 17 lines
  
  sip codec negotiation of dynamic rtp payloads error fix
  
  This patch fixes how chan_sip handles dynamic rtp payload types
  it does not understand.  At the moment if a dynamic payload's mime
  type does not match one we understand, the payload does not get
  removed from our payload table.  As a result of this, the payload
  is set to whatever dynamic codec we use internally for that payload
  number on outgoing INVITES.  This is incorrect.
  
  This patch fixes this by properly checking the rtpmap set function's
  return code to make sure it was found.  The function can return both
  -1 and -2 depending on the source of the mismatch.  We were just
  checking -1 explicitly.
  
  Review: https://reviewboard.asterisk.org/r/1169/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314018 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
David Vossel
2011-04-18 13:42:51 +00:00
parent 6b01eb3324
commit 4b4549106b
3 changed files with 7 additions and 6 deletions
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6
channels/chan_sip.c
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@@ -9092,8 +9092,8 @@ static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp_
} else if (sscanf(a, "rtpmap: %30u %127[^/]/%30u", &codec, mimeSubtype, &sample_rate) == 3) {
/* We have a rtpmap to handle */
if (*last_rtpmap_codec < SDP_MAX_RTPMAP_CODECS) {
if (ast_rtp_codecs_payloads_set_rtpmap_type_rate(newaudiortp, NULL, codec, "audio", mimeSubtype,
ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0, sample_rate) != -1) {
if (!(ast_rtp_codecs_payloads_set_rtpmap_type_rate(newaudiortp, NULL, codec, "audio", mimeSubtype,
ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0, sample_rate))) {
if (debug)
ast_verbose("Found audio description format %s for ID %d\n", mimeSubtype, codec);
//found_rtpmap_codecs[last_rtpmap_codec] = codec;
@@ -9179,7 +9179,7 @@ static int process_sdp_a_video(const char *a, struct sip_pvt *p, struct ast_rtp_
if (*last_rtpmap_codec < SDP_MAX_RTPMAP_CODECS) {
/* Note: should really look at the '#chans' params too */
if (!strncasecmp(mimeSubtype, "H26", 3) || !strncasecmp(mimeSubtype, "MP4", 3)) {
if (ast_rtp_codecs_payloads_set_rtpmap_type_rate(newvideortp, NULL, codec, "video", mimeSubtype, 0, sample_rate) != -1) {
if (!(ast_rtp_codecs_payloads_set_rtpmap_type_rate(newvideortp, NULL, codec, "video", mimeSubtype, 0, sample_rate))) {
if (debug)
ast_verbose("Found video description format %s for ID %d\n", mimeSubtype, codec);
//found_rtpmap_codecs[last_rtpmap_codec] = codec;