chan_sip: Fix autoframing=yes.
authorAlexander Traud <pabstraud@compuserve.com>
Wed, 21 Oct 2015 14:51:11 +0000 (16:51 +0200)
committerAlexander Traud <pabstraud@compuserve.com>
Wed, 21 Oct 2015 14:51:55 +0000 (16:51 +0200)
With Asterisk 13, the structures ast_format and ast_codec changed. Because of
that, the paketization timing (framing) of the RTP channel moved away from the
formats/codecs. In the course of that change, the ptime of the callee was not
honored anymore, when the optional autoframing was enabled.

ASTERISK-25484 #close

Change-Id: Ic600ccaa125e705922f89c72212c698215d239b4

channels/chan_sip.c

index 3fdc3ca..8f76e9c 100644 (file)
@@ -11095,7 +11095,7 @@ 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_rtp_codecs_set_framing(ast_rtp_instance_get_codecs(p->rtp), framing);
+                       ast_format_cap_set_framing(p->caps, framing);
                }
                found = TRUE;
        } else if (sscanf(a, "rtpmap: %30u %127[^/]/%30u", &codec, mimeSubtype, &sample_rate) == 3) {
@@ -13384,6 +13384,11 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
                        ast_str_append(&a_audio, 0, "a=maxptime:%d\r\n", max_audio_packet_size);
                }
 
+               if (!ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
+                       ast_debug(1, "Setting framing on incoming call: %u\n", min_audio_packet_size);
+                       ast_rtp_codecs_set_framing(ast_rtp_instance_get_codecs(p->rtp), min_audio_packet_size);
+               }
+
                if (!doing_directmedia) {
                        if (ast_test_flag(&p->flags[2], SIP_PAGE3_ICE_SUPPORT)) {
                                add_ice_to_sdp(p->rtp, &a_audio);
@@ -13676,10 +13681,6 @@ static int transmit_response_with_sdp(struct sip_pvt *p, const char *msg, const
                add_cc_call_info_to_response(p, &resp);
        }
        if (p->rtp) {
-               if (!p->autoframing && !ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
-                       ast_debug(1, "Setting framing from config on incoming call\n");
-                       ast_rtp_codecs_set_framing(ast_rtp_instance_get_codecs(p->rtp), ast_format_cap_get_framing(p->caps));
-               }
                ast_rtp_instance_activate(p->rtp);
                try_suggested_sip_codec(p);
                if (p->t38.state == T38_ENABLED) {