kenmirrors/asterisk
1
0
Fork 0
mirror of https://github.com/asterisk/asterisk.git synced 2025-09-03 11:25:35 +00:00
Code Issues Packages Projects Releases Wiki Activity

chan_sip: Handle invalid SDP answer to T.38 re-invite

Browse Source 
The chan_sip module performs a T.38 re-invite using a single media
stream of udptl, and expects the SDP answer to be the same.

If an SDP answer is received instead that contains an additional
media stream with no joint codec a crash will occur as the code
assumes that at least one joint codec will exist in this
scenario.

This change removes this assumption.

ASTERISK-28465

Change-Id: I8b02845b53344c6babe867a3f0a5231045c7ac87
This commit is contained in:
Francesco Castellano
2019-06-28 18:15:31 +02:00
committed by Gerrit
parent 99addaff69
commit 8438d19b81
1 changed files with 7 additions and 1 deletions
Download Patch File Download Diff File Expand all files Collapse all files

8
channels/chan_sip.c
View File

@@ -10965,7 +10965,13 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
ast_rtp_lookup_mime_multiple2(s3, NULL, newnoncodeccapability, 0, 0));
}
if (portno != -1 || vportno != -1 || tportno != -1) {
/* When UDPTL is negotiated it is expected that there are no compatible codecs as audio or
* video is not being transported, thus we continue in this function further up if that is
* the case. If we receive an SDP answer containing both a UDPTL stream and another media
* stream however we need to check again to ensure that there is at least one joint codec
* instead of assuming there is one.
*/
if ((portno != -1 || vportno != -1 || tportno != -1) && ast_format_cap_count(newjointcapability)) {
/* We are now ready to change the sip session and RTP structures with the offered codecs, since
they are acceptable */
unsigned int framing;