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asterisk
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Don't count unknown media streams as media streams in the offer... (Fenlander reported this on irc)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@35095 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.4
Olle Johansson 20 years ago
parent e04914b44a
commit cb4221c7e9
1 changed files with 4 additions and 2 deletions
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6
channels/chan_sip.c
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@ -4506,7 +4506,6 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
while ((m = get_sdp_iterate(&iterator, req, "m"))[0] != '\0') { while ((m = get_sdp_iterate(&iterator, req, "m"))[0] != '\0') {
int x; int x;
int audio = FALSE; int audio = FALSE;
numberofmediastreams++;
if (p->vrtp) if (p->vrtp)
ast_rtp_pt_clear(newvideortp); /* Must be cleared in case no m=video line exists */ ast_rtp_pt_clear(newvideortp); /* Must be cleared in case no m=video line exists */
@ -4514,6 +4513,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
if ((sscanf(m, "audio %d/%d RTP/AVP %n", &x, &numberofports, &len) == 2) || if ((sscanf(m, "audio %d/%d RTP/AVP %n", &x, &numberofports, &len) == 2) ||
(sscanf(m, "audio %d RTP/AVP %n", &x, &len) == 1)) { (sscanf(m, "audio %d RTP/AVP %n", &x, &len) == 1)) {
audio = TRUE; audio = TRUE;
numberofmediastreams++;
/* Found audio stream in this media definition */ /* Found audio stream in this media definition */
portno = x; portno = x;
/* Scan through the RTP payload types specified in a "m=" line: */ /* Scan through the RTP payload types specified in a "m=" line: */
@ -4531,6 +4531,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
(sscanf(m, "video %d RTP/AVP %n", &x, &len) == 1)) { (sscanf(m, "video %d RTP/AVP %n", &x, &len) == 1)) {
/* If it is not audio - is it video ? */ /* If it is not audio - is it video ? */
ast_clear_flag(&p->flags[0], SIP_NOVIDEO); ast_clear_flag(&p->flags[0], SIP_NOVIDEO);
numberofmediastreams++;
vportno = x; vportno = x;
/* Scan through the RTP payload types specified in a "m=" line: */ /* Scan through the RTP payload types specified in a "m=" line: */
for (codecs = m + len; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) { for (codecs = m + len; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) {
@ -4546,6 +4547,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
if (debug) if (debug)
ast_verbose("Got T.38 offer in SDP in dialog %s\n", p->callid); ast_verbose("Got T.38 offer in SDP in dialog %s\n", p->callid);
udptlportno = x; udptlportno = x;
numberofmediastreams++;
if (p->owner && p->lastinvite) { if (p->owner && p->lastinvite) {
p->t38.state = T38_PEER_REINVITE; /* T38 Offered in re-invite from remote party */ p->t38.state = T38_PEER_REINVITE; /* T38 Offered in re-invite from remote party */
@ -4584,7 +4586,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
return -2; return -2;
if (numberofmediastreams > 2) if (numberofmediastreams > 2)
/* We have too many media streams, fail this offer */ /* We have too many fax, audio and/or video media streams, fail this offer */
return -3; return -3;
/* RTP addresses and ports for audio and video */ /* RTP addresses and ports for audio and video */

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