[Asterisk-cvs] asterisk/channels chan_sip.c,1.546,1.547
markster at lists.digium.com
markster at lists.digium.com
Sun Oct 31 20:58:57 CST 2004
Update of /usr/cvsroot/asterisk/channels
In directory mongoose.digium.com:/tmp/cvs-serv17575/channels
Modified Files:
chan_sip.c
Log Message:
Accept %d/%d in m= line of SDP (bug #2625, but not their patch)
Index: chan_sip.c
===================================================================
RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v
retrieving revision 1.546
retrieving revision 1.547
diff -u -d -r1.546 -r1.547
--- chan_sip.c 28 Oct 2004 15:53:36 -0000 1.546
+++ chan_sip.c 1 Nov 2004 02:01:37 -0000 1.547
@@ -2607,7 +2607,7 @@
int codec;
int iterator;
int sendonly = 0;
- int x;
+ int x,y;
int debug=sip_debug_test_pvt(p);
/* Update our last rtprx when we receive an SDP, too */
@@ -2637,7 +2637,8 @@
sdpLineNum_iterator_init(&iterator);
p->novideo = 1;
while ((m = get_sdp_iterate(&iterator, req, "m"))[0] != '\0') {
- if ((sscanf(m, "audio %d RTP/AVP %n", &x, &len) == 1)) {
+ if ((sscanf(m, "audio %d RTP/AVP %n", &x, &len) == 1) ||
+ (sscanf(m, "audio %d/%d RTP/AVP %n", &x, &y, &len) == 1)) {
portno = x;
/* Scan through the RTP payload types specified in a "m=" line: */
ast_rtp_pt_clear(p->rtp);
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