kenmirrors/asterisk
1
0
Fork 0
mirror of https://github.com/asterisk/asterisk.git synced 2025-09-05 12:16:00 +00:00
Code Issues Packages Projects Releases Wiki Activity

Merge "chan_sip: Fix improper RTP framing on outgoing calls"

Browse Source
This commit is contained in:
Jenkins2
2018-03-08 15:53:35 -06:00
committed by Gerrit Code Review
parent 257be6847c 75a35ee5e8
commit caed6671a3
1 changed files with 9 additions and 5 deletions
Download Patch File Download Diff File Expand all files Collapse all files

14
channels/chan_sip.c
View File

@@ -10964,22 +10964,25 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
if (portno != -1 || vportno != -1 || tportno != -1) {
/* We are now ready to change the sip session and RTP structures with the offered codecs, since
they are acceptable */
unsigned int framing;
ast_format_cap_remove_by_type(p->jointcaps, AST_MEDIA_TYPE_UNKNOWN);
ast_format_cap_append_from_cap(p->jointcaps, newjointcapability, AST_MEDIA_TYPE_UNKNOWN); /* Our joint codec profile for this call */
ast_format_cap_remove_by_type(p->peercaps, AST_MEDIA_TYPE_UNKNOWN);
ast_format_cap_append_from_cap(p->peercaps, newpeercapability, AST_MEDIA_TYPE_UNKNOWN); /* The other side's capability in latest offer */
p->jointnoncodeccapability = newnoncodeccapability; /* DTMF capabilities */
tmp_fmt = ast_format_cap_get_format(p->jointcaps, 0);
framing = ast_format_cap_get_format_framing(p->jointcaps, tmp_fmt);
/* respond with single most preferred joint codec, limiting the other side's choice */
if (ast_test_flag(&p->flags[1], SIP_PAGE2_PREFERRED_CODEC)) {
unsigned int framing;
tmp_fmt = ast_format_cap_get_format(p->jointcaps, 0);
framing = ast_format_cap_get_format_framing(p->jointcaps, tmp_fmt);
ast_format_cap_remove_by_type(p->jointcaps, AST_MEDIA_TYPE_UNKNOWN);
ast_format_cap_append(p->jointcaps, tmp_fmt, framing);
ao2_ref(tmp_fmt, -1);
}
if (!ast_rtp_codecs_get_framing(&newaudiortp)) {
/* Peer did not force us to use a specific framing, so use our own */
ast_rtp_codecs_set_framing(&newaudiortp, framing);
}
ao2_ref(tmp_fmt, -1);
}
/* Setup audio address and port */
@@ -11488,6 +11491,7 @@ static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp_
if (framing && p->autoframing) {
ast_debug(1, "Setting framing to %ld\n", framing);
ast_format_cap_set_framing(p->caps, framing);
ast_rtp_codecs_set_framing(newaudiortp, framing);
}
found = TRUE;
} else if (sscanf(a, "rtpmap: %30u %127[^/]/%30u", &codec, mimeSubtype, &sample_rate) == 3) {