kenmirrors/asterisk
1
0
Fork 0
mirror of https://github.com/asterisk/asterisk.git synced 2025-09-22 13:25:17 +00:00
Code Issues Packages Projects Releases Wiki Activity

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

Browse Source 
- 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
This commit is contained in:
Olle Johansson
2006-11-12 15:30:48 +00:00
parent 1f9dcf758c
commit b744a76565
1 changed files with 26 additions and 19 deletions
Download Patch File Download Diff File Expand all files Collapse all files

45
channels/chan_sip.c
View File

@@ -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*/