Merge "chan_sip: Fix improper RTP framing on outgoing calls"
authorJenkins2 <jenkins2@gerrit.asterisk.org>
Thu, 8 Mar 2018 21:53:35 +0000 (15:53 -0600)
committerGerrit Code Review <gerrit2@gerrit.digium.api>
Thu, 8 Mar 2018 21:53:35 +0000 (15:53 -0600)
channels/chan_sip.c

index 8570163..f0cc2a6 100644 (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) {