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

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 5c28cf29502b1c6351e704622f5f8f549f260e4f..b42e62557b2597f619acfed6bbcc2e36259b5361 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -10525,7 +10525,11 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
        /* 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) {
@@ -13271,6 +13275,11 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
                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);