Fix regression introduced by SDP fixups

If capability is adjusted when switching to UDPTL during fax transmission, fax
teardown fails.  Make sure capability is only touched if RTP is active.  This
regression was introduced in R344385.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@344769 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index bdf948ace8ca0534382aadf732bb1aecde71d1e0..aaf71567e26b996bcf34dc4a3620984a370351e8 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -9093,15 +9093,17 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
                return -1;
        }
 
-       /* We are now ready to change the sip session and p->rtp and p->vrtp with the offered codecs, since
-          they are acceptable */
-       p->jointcapability = newjointcapability;                /* Our joint codec profile for this call */
-       p->peercapability = newpeercapability;                  /* The other sides capability in latest offer */
-       p->jointnoncodeccapability = newnoncodeccapability;     /* DTMF capabilities */
-
-       /* respond with single most preferred joint codec, limiting the other side's choice */
-       if (ast_test_flag(&p->flags[1], SIP_PAGE2_PREFERRED_CODEC)) {
-               p->jointcapability = ast_codec_choose(&p->prefs, p->jointcapability, 1);
+       if (portno != -1 || vportno != -1 || tportno != -1) {
+               /* We are now ready to change the sip session and p->rtp and p->vrtp with the offered codecs, since
+                  they are acceptable */
+               p->jointcapability = newjointcapability;                /* Our joint codec profile for this call */
+               p->peercapability = newpeercapability;                  /* The other sides capability in latest offer */
+               p->jointnoncodeccapability = newnoncodeccapability;     /* DTMF capabilities */
+       
+               /* respond with single most preferred joint codec, limiting the other side's choice */
+               if (ast_test_flag(&p->flags[1], SIP_PAGE2_PREFERRED_CODEC)) {
+                       p->jointcapability = ast_codec_choose(&p->prefs, p->jointcapability, 1);
+               }
        }
 
        /* Setup audio address and port */