Merged revisions 162188 via svnmerge from

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r162188 | file | 2008-12-09 15:06:14 -0400 (Tue, 09 Dec 2008) | 4 lines

  Take video into account when early bridging RTP.
  (closes issue #13535)
  Reported by: davidw
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162197 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/main/rtp.c b/main/rtp.c
index 9f4c342..1cf482e 100644 (file)
--- a/main/rtp.c
+++ b/main/rtp.c
@@ -2056,18 +2056,18 @@ int ast_rtp_early_bridge(struct ast_channel *c0, struct ast_channel *c1)
        }
 
        /* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
        }
 
        /* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */

-       if (audio_dest_res != AST_RTP_TRY_NATIVE) {
+       if (audio_dest_res != AST_RTP_TRY_NATIVE || (video_dest_res != AST_RTP_GET_FAILED && video_dest_res != AST_RTP_TRY_NATIVE)) {

                /* Somebody doesn't want to play... */
                ast_channel_unlock(c0);
                if (c1)
                        ast_channel_unlock(c1);
                return -1;
        }
                /* Somebody doesn't want to play... */
                ast_channel_unlock(c0);
                if (c1)
                        ast_channel_unlock(c1);
                return -1;
        }

-       if (audio_src_res == AST_RTP_TRY_NATIVE && srcpr->get_codec)
+       if (audio_src_res == AST_RTP_TRY_NATIVE && (video_src_res == AST_RTP_GET_FAILED || video_src_res == AST_RTP_TRY_NATIVE) && srcpr->get_codec)

                srccodec = srcpr->get_codec(c1);
        else
                srccodec = 0;
                srccodec = srcpr->get_codec(c1);
        else
                srccodec = 0;

-       if (audio_dest_res == AST_RTP_TRY_NATIVE && destpr->get_codec)
+       if (audio_dest_res == AST_RTP_TRY_NATIVE && (video_dest_res == AST_RTP_GET_FAILED || video_dest_res == AST_RTP_TRY_NATIVE) && destpr->get_codec)

                destcodec = destpr->get_codec(c0);
        else
                destcodec = 0;
                destcodec = destpr->get_codec(c0);
        else
                destcodec = 0;

@@ -2144,7 +2144,7 @@ int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src, i
                destcodec = 0;
 
        /* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
                destcodec = 0;
 
        /* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */

-       if (audio_dest_res != AST_RTP_TRY_NATIVE || audio_src_res != AST_RTP_TRY_NATIVE || !(srccodec & destcodec)) {
+       if (audio_dest_res != AST_RTP_TRY_NATIVE || (video_dest_res != AST_RTP_GET_FAILED && video_dest_res != AST_RTP_TRY_NATIVE) || audio_src_res != AST_RTP_TRY_NATIVE || (video_src_res != AST_RTP_GET_FAILED && video_src_res != AST_RTP_TRY_NATIVE) || !(srccodec & destcodec)) {

                /* Somebody doesn't want to play... */
                ast_channel_unlock(dest);
                ast_channel_unlock(src);
                /* Somebody doesn't want to play... */
                ast_channel_unlock(dest);
                ast_channel_unlock(src);