Merged revisions 129436 via svnmerge from

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

........
r129436 | mmichelson | 2008-07-09 14:32:20 -0500 (Wed, 09 Jul 2008) | 13 lines

Fix a problem where inbound rfc2833 audio would be sent to the
core instead of being P2P bridged. When the core regenerated
the rfc2833 packet for the outbound leg, the SSRC would be different
than the RTP audio on the call leg causing DTMF detection issues on
the far end.

(closes issue #12955)
Reported by: tonyredstone
Patches:
      dynamic_rtp.patch uploaded by tsearle (license 373)
Tested by: tonyredstone

........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129437 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/main/rtp.c b/main/rtp.c
index 1370e12d28ea94dda0c0fe3bbdc54885ab205772..bc3db9b4b6a041fc0b1511492922cb097c21b1cc 100644 (file)
--- a/main/rtp.c
+++ b/main/rtp.c
@@ -1479,10 +1479,6 @@ static int bridge_p2p_rtp_write(struct ast_rtp *rtp, struct ast_rtp *bridged, un
        /* Check what the payload value should be */
        rtpPT = ast_rtp_lookup_pt(rtp, payload);
 
-       /* If the payload coming in is not one of the negotiated ones then send it to the core, this will cause formats to change and the bridge to break */
-       if (!bridged->current_RTP_PT[payload].code)
-               return -1;
-
        /* If the payload is DTMF, and we are listening for DTMF - then feed it into the core */
        if (ast_test_flag(rtp, FLAG_P2P_NEED_DTMF) && !rtpPT.isAstFormat && rtpPT.code == AST_RTP_DTMF)
                return -1;
@@ -1490,6 +1486,11 @@ static int bridge_p2p_rtp_write(struct ast_rtp *rtp, struct ast_rtp *bridged, un
        /* Otherwise adjust bridged payload to match */
        bridged_payload = ast_rtp_lookup_code(bridged, rtpPT.isAstFormat, rtpPT.code);
 
+       /* If the payload coming in is not one of the negotiated ones then send it to the core, this will cause formats to change and the bridge to break */
+       if (!bridged->current_RTP_PT[bridged_payload].code)
+               return -1;
+
+
        /* If the mark bit has not been sent yet... do it now */
        if (!ast_test_flag(rtp, FLAG_P2P_SENT_MARK)) {
                mark = 1;