[asterisk-commits] chan sip: Fix autoframing=yes. (asterisk[master])

[SVN commits to the Asterisk project]

Joshua Colp has submitted this change and it was merged.

Change subject: chan_sip: Fix autoframing=yes.
......................................................................


chan_sip: Fix autoframing=yes.

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
---
M channels/chan_sip.c
1 file changed, 6 insertions(+), 5 deletions(-)

Approvals:
  Kevin Harwell: Looks good to me, but someone else must approve
  Anonymous Coward #1000019: Verified
  Joshua Colp: Looks good to me, approved



diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 3fdc3ca..8f76e9c 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -11095,7 +11095,7 @@
 
 		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 @@
 			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 @@
 		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) {

-- 
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Gerrit-MessageType: merged
Gerrit-Change-Id: Ic600ccaa125e705922f89c72212c698215d239b4
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Owner: Alexander Traud <pabstraud at compuserve.com>
Gerrit-Reviewer: Alexander Traud <pabstraud at compuserve.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Kevin Harwell <kharwell at digium.com>