- Don't hangup because of failed re-invite. Go back to previous state.

- Keep RTP running during T.38 session
  We might improve the code to issue ast_rtp_stop if T.38 re-invite not fails
  later on in the code, but I don't see many reasons to.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47510 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index a206f73496ddbf47533de253888b03d0d7bedbc9..e7c552c3e43816cb5b32ac35e40ad638c1af4602 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -4862,22 +4862,6 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
        memcpy(&sin.sin_addr, hp->h_addr, sizeof(sin.sin_addr));
        if (vhp)
                memcpy(&vsin.sin_addr, vhp->h_addr, sizeof(vsin.sin_addr));
-               
-       if (p->rtp) {
-               if (portno > 0) {
-                       sin.sin_port = htons(portno);
-                       ast_rtp_set_peer(p->rtp, &sin);
-                       if (debug)
-                               ast_verbose("Peer audio RTP is at port %s:%d\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
-               } else {
-                       ast_rtp_stop(p->rtp);
-                       if (debug)
-                               ast_verbose("Peer doesn't provide audio\n");
-               }
-       }
-       /* Setup video port number */
-       if (vportno != -1)
-               vsin.sin_port = htons(vportno);
 
        /* Setup UDPTL port number */
        if (p->udptl) {
@@ -4893,6 +4877,28 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
                }
        }
 
+               
+       if (p->rtp) {
+               if (portno > 0) {
+                       sin.sin_port = htons(portno);
+                       ast_rtp_set_peer(p->rtp, &sin);
+                       if (debug)
+                               ast_verbose("Peer audio RTP is at port %s:%d\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
+               } else {
+                       if (udptlportno > 0) {
+                               if (debug)
+                                       ast_verbose("Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid %s\n", p->callid);
+                       } else {
+                               ast_rtp_stop(p->rtp);
+                               if (debug)
+                                       ast_verbose("Peer doesn't provide audio. Callid %s\n", p->callid);
+                       }
+               }
+       }
+       /* Setup video port number */
+       if (vportno != -1)
+               vsin.sin_port = htons(vportno);
+
        /* Next, scan through each "a=rtpmap:" line, noting each
         * specified RTP payload type (with corresponding MIME subtype):
         */
@@ -13479,7 +13485,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
                                                                        transmit_response(p, "488 Not acceptable here", req);
                                                                else
                                                                        transmit_response_reliable(p, "488 Not acceptable here", req);
-                                                               sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
+                                                       
                                                        }
                                                } else {
                                                        /* The other side is already setup for T.38 most likely so we need to acknowledge this too */
@@ -13497,7 +13503,9 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
                                                p->t38.state = T38_DISABLED;
                                                if (option_debug > 1)
                                                        ast_log(LOG_DEBUG,"T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>");
-                                               sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
+
+                                               if (!p->lastinvite) /* Only destroy if this is *not* a re-invite */
+                                                       sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
                                        }
                                } else {
                                        /* we are not bridged in a call */
@@ -13524,7 +13532,6 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
                                                                transmit_response(p, "488 Not Acceptable Here (unsupported)", req);
                                                        else
                                                                transmit_response_reliable(p, "488 Not Acceptable Here (unsupported)", req);
-                                                       sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
                                                        sendok = FALSE;
                                                } 
                                                /* No bridged peer with T38 enabled*/