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Start out with cleared RTP payload structures instead of defaults. This should prevent issues where if a stream (audio/stream) is not present and it's RTP payload structure is combined with the overall capability then the capability would be every codec that Asterisk supports.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38158 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Joshua Colp
2006-07-24 15:47:59 +00:00
parent 2c3bc8b1b3
commit cbf79ca489
1 changed files with 4 additions and 9 deletions
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13
channels/chan_sip.c
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@@ -4478,11 +4478,11 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
/* Initialize the temporary RTP structures we use to evaluate the offer from the peer */
newaudiortp = alloca(ast_rtp_alloc_size());
memset(newaudiortp, 0, ast_rtp_alloc_size());
ast_rtp_pt_default(newaudiortp);
ast_rtp_pt_clear(newaudiortp);
newvideortp = alloca(ast_rtp_alloc_size());
memset(newvideortp, 0, ast_rtp_alloc_size());
ast_rtp_pt_default(newvideortp);
ast_rtp_pt_clear(newvideortp);
/* Update our last rtprx when we receive an SDP, too */
p->lastrtprx = p->lastrtptx = time(NULL); /* XXX why both ? */
@@ -4520,8 +4520,6 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
int x;
int audio = FALSE;
if (p->vrtp)
ast_rtp_pt_clear(newvideortp); /* Must be cleared in case no m=video line exists */
numberofports = 1;
if ((sscanf(m, "audio %d/%d RTP/AVP %n", &x, &numberofports, &len) == 2) ||
(sscanf(m, "audio %d RTP/AVP %n", &x, &len) == 1)) {
@@ -4530,7 +4528,6 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
/* Found audio stream in this media definition */
portno = x;
/* Scan through the RTP payload types specified in a "m=" line: */
ast_rtp_pt_clear(newaudiortp);
for (codecs = m + len; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) {
if (sscanf(codecs, "%d%n", &codec, &len) != 1) {
ast_log(LOG_WARNING, "Error in codec string '%s'\n", codecs);
@@ -4825,10 +4822,8 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
}
/* Now gather all of the codecs that we are asked for: */
if (p->rtp)
ast_rtp_get_current_formats(newaudiortp, &peercapability, &peernoncodeccapability);
if (p->vrtp)
ast_rtp_get_current_formats(newvideortp, &vpeercapability, &vpeernoncodeccapability);
ast_rtp_get_current_formats(newaudiortp, &peercapability, &peernoncodeccapability);
ast_rtp_get_current_formats(newvideortp, &vpeercapability, &vpeernoncodeccapability);
newjointcapability = p->capability & (peercapability | vpeercapability);
newpeercapability = (peercapability | vpeercapability);