aboutsummaryrefslogtreecommitdiffstats
path: root/channels
diff options
context:
space:
mode:
authoroej <oej@f38db490-d61c-443f-a65b-d21fe96a405b>2006-11-12 15:27:00 +0000
committeroej <oej@f38db490-d61c-443f-a65b-d21fe96a405b>2006-11-12 15:27:00 +0000
commit696dca54672a5c9f897489439145f578c1ae8816 (patch)
treefca0195b472665f4746d7809594116c638c35bea /channels
parent608754a18fd7521e4f37bfe2a735aaa7c6aefe58 (diff)
- Don't destroy SIP dialog because of a failed T.38 re-invite. Wait for a bye.
Final response to a re-invite does not mean that the session dies, only that the re-invite fails. - Keep RTP active during processing of T.38 re-invite. If the re-invite fails, RTP needs to remain as before the re-invite. Issue 8338 - darren1713. Please test. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@47509 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'channels')
-rw-r--r--channels/chan_sip.c45
1 files changed, 26 insertions, 19 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 059366896..2e1fe220c 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -4764,22 +4764,6 @@ 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) {
@@ -4795,6 +4779,28 @@ 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):
*/
@@ -13245,7 +13251,7 @@ 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 */
@@ -13263,7 +13269,9 @@ 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 */
@@ -13290,7 +13298,6 @@ 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*/