chan_sip: Fix improper RTP framing on outgoing calls

The "ptime" SDP parameter received in a SIP response was not honoured.
Moreover, in the abscence of this "ptime" parameter, locally configured
framing was lost during response processing.

This patch systematically stores the framing information in the
ast_rtp_codecs structure, taking it from the response or from the
configuration as appropriate.

ASTERISK-27674

Change-Id: I828a6a98d27a45a8afd07236a2bd0aa3cbd3fb2c

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index a24ca81152cb054334e4854051a3f5ba1a43dbbd..76848c1dd5313bafdd5f2e3546bbcd552393e41b 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -10920,22 +10920,25 @@ 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 */
@@ -11444,6 +11447,7 @@ 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) {