Start out with cleared RTP payload structures instead of defaults. This should preven...

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38158 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 9713021..f3ff615 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -4478,11 +4478,11 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
        /* Initialize the temporary RTP structures we use to evaluate the offer from the peer */
        newaudiortp = alloca(ast_rtp_alloc_size());
        memset(newaudiortp, 0, ast_rtp_alloc_size());
-       ast_rtp_pt_default(newaudiortp);
+       ast_rtp_pt_clear(newaudiortp);
 
        newvideortp = alloca(ast_rtp_alloc_size());
        memset(newvideortp, 0, ast_rtp_alloc_size());
-       ast_rtp_pt_default(newvideortp);
+       ast_rtp_pt_clear(newvideortp);
 
        /* Update our last rtprx when we receive an SDP, too */
        p->lastrtprx = p->lastrtptx = time(NULL); /* XXX why both ? */
@@ -4520,8 +4520,6 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
                int x;
                int audio = FALSE;
 
-               if (p->vrtp)
-                       ast_rtp_pt_clear(newvideortp);  /* Must be cleared in case no m=video line exists */
                numberofports = 1;
                if ((sscanf(m, "audio %d/%d RTP/AVP %n", &x, &numberofports, &len) == 2) ||
                    (sscanf(m, "audio %d RTP/AVP %n", &x, &len) == 1)) {
@@ -4530,7 +4528,6 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
                        /* Found audio stream in this media definition */
                        portno = x;
                        /* Scan through the RTP payload types specified in a "m=" line: */
-                       ast_rtp_pt_clear(newaudiortp);
                        for (codecs = m + len; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) {
                                if (sscanf(codecs, "%d%n", &codec, &len) != 1) {
                                        ast_log(LOG_WARNING, "Error in codec string '%s'\n", codecs);
@@ -4825,10 +4822,8 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
        }
 
        /* Now gather all of the codecs that we are asked for: */
-       if (p->rtp)
-               ast_rtp_get_current_formats(newaudiortp, &peercapability, &peernoncodeccapability);
-       if (p->vrtp)
-               ast_rtp_get_current_formats(newvideortp, &vpeercapability, &vpeernoncodeccapability);
+       ast_rtp_get_current_formats(newaudiortp, &peercapability, &peernoncodeccapability);
+       ast_rtp_get_current_formats(newvideortp, &vpeercapability, &vpeernoncodeccapability);
 
        newjointcapability = p->capability & (peercapability | vpeercapability);
        newpeercapability = (peercapability | vpeercapability);