[asterisk-commits] chan sip.c: Start ICE negotiation when response is sent or r... (asterisk[13])

[SVN commits to the Asterisk project]

Joshua Colp has submitted this change and it was merged.

Change subject: chan_sip.c: Start ICE negotiation when response is sent or received.
......................................................................


chan_sip.c: Start ICE negotiation when response is sent or received.

The current logic for ICE negotiation starts it
when receiving an SDP with ICE candidates. This is
incorrect as ICE negotiation can only start when each 
call party have at least one pair of local and remote 
candidate. Starting ICE negotiation early would result 
in negotiation failure and ultimately no audio.

This change makes it so ICE negotiation is only started
when a response with SDP is received or when a response
with SDP is sent.

ASTERISK-24146

Change-Id: I55a632bde9e9827871b09141d82747e08379a8ca
---
M channels/chan_sip.c
1 file changed, 10 insertions(+), 1 deletion(-)

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 47b505d..c542245 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -10628,7 +10628,11 @@
 	/* Setup audio address and port */
 	if (p->rtp) {
 		if (sa && portno > 0) {
-			start_ice(p->rtp, (req->method != SIP_RESPONSE) ? 0 : 1);
+			/* Start ICE negotiation here, only when it is response, and setting that we are conrolling agent,
+			   as we are offerer */
+			if (req->method == SIP_RESPONSE) {
+				start_ice(p->rtp, 1);
+			}
 			ast_sockaddr_set_port(sa, portno);
 			ast_rtp_instance_set_remote_address(p->rtp, sa);
 			if (debug) {
@@ -13402,6 +13406,11 @@
 		if (!doing_directmedia) {
 			if (ast_test_flag(&p->flags[2], SIP_PAGE3_ICE_SUPPORT)) {
 				add_ice_to_sdp(p->rtp, &a_audio);
+				/* Start ICE negotiation, and setting that we are controlled agent,
+				   as this is response to offer */
+				if (resp->method == SIP_RESPONSE) {
+					start_ice(p->rtp, 0);
+				}
 			}
 
 			add_dtls_to_sdp(p->rtp, &a_audio);

-- 
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Gerrit-MessageType: merged
Gerrit-Change-Id: I55a632bde9e9827871b09141d82747e08379a8ca
Gerrit-PatchSet: 2
Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-Owner: Eugene Voityuk <eugene at thirdlane.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Kevin Harwell <kharwell at digium.com>