Add another field to track whether video was invited or not (bug #2226)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@3619 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index b59d4f5e1bd7e9d535c63e9a0e34a66c15b6119e..bb0a100bedb86e96aaf29dc480975a1448ef0109 100755 (executable)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -262,6 +262,7 @@ static struct sip_pvt {
        int alreadygone;                        /* Whether or not we've already been destroyed by or peer */
        int needdestroy;                        /* if we need to be destroyed */
        int capability;                         /* Special capability (codec) */
+       int novideo;                            /* Didn't get video in invite, don't offer */
        int jointcapability;                    /* Supported capability at both ends (codecs ) */
        int prefcodec;                          /* Preferred codec (outbound only) */
        int noncodeccapability;
@@ -2639,6 +2640,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
                return -1;
        }
        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)) {
                        portno = x;
@@ -2662,6 +2664,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
                        ast_rtp_pt_clear(p->vrtp);  /* Must be cleared in case no m=video line exists */
 
                if (p->vrtp && (sscanf(m, "video %d RTP/AVP %n", &x, &len) == 1)) {
+                       p->novideo = 0;
                        vportno = x;
                        /* Scan through the RTP payload types specified in a "m=" line: */
                        codecs = m + len;
@@ -3450,7 +3453,7 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p)
        if ((sizeof(m) <= strlen(m) - 2) || (sizeof(m2) <= strlen(m2) - 2) || (sizeof(a) == strlen(a)) || (sizeof(a2) == strlen(a2)))
                ast_log(LOG_WARNING, "SIP SDP may be truncated due to undersized buffer!!\n");
        len = strlen(v) + strlen(s) + strlen(o) + strlen(c) + strlen(t) + strlen(m) + strlen(a);
-       if ((p->vrtp) && (capability & VIDEO_CODEC_MASK)) /* only if video response is appropriate */
+       if ((p->vrtp) && (!p->novideo) && (capability & VIDEO_CODEC_MASK)) /* only if video response is appropriate */
                len += strlen(m2) + strlen(a2);
        snprintf(costr, sizeof(costr), "%d", len);
        add_header(resp, "Content-Type", "application/sdp");
@@ -3462,7 +3465,7 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p)
        add_line(resp, t);
        add_line(resp, m);
        add_line(resp, a);
-       if ((p->vrtp) && (capability & VIDEO_CODEC_MASK)) { /* only if video response is appropriate */
+       if ((p->vrtp) && (!p->novideo) && (capability & VIDEO_CODEC_MASK)) { /* only if video response is appropriate */
                add_line(resp, m2);
                add_line(resp, a2);
        }