[asterisk-bugs] [Asterisk 0017007]: [patch] RTP Timestamp changes after transfer, but SSRC not and the markerbit ist not set.

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=17007 
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Reported By:                addix
Assigned To:                twilson
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Project:                    Asterisk
Issue ID:                   17007
Category:                   Channels/chan_sip/Transfers
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     ready for review
Target Version:             1.6.2.11
Asterisk Version:           SVN 
JIRA:                       SWP-1096 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2010-03-11 10:07 CST
Last Modified:              2010-07-19 03:35 CDT
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Summary:                    [patch] RTP Timestamp changes after transfer, but
SSRC not and the markerbit ist not set.
Description: 
On every SIP Transfer (Example: A calls B / B places A on hold / B calls C
/ A sends Transfer to Asterisk PBX) the Outing RTP Traffic from Asterisk to
the transfer target (RTP to C) is broken. The Asterisk is changing the RTP
Timestamp massivly but the SSRC stays on the old value and the timestamp
marker is also not set. As soon as the new timestamp is smaller than the
old timestamp value the transfer target rejects the RTP Packets after the
transfer (Not really, it's just not played), so i get one way audio.

I experienced that with serveral local SIP-Carriers and Funkwerk Rxxxx
BRI/PRI Mediagateways as transfer target.

Due to my limited Asterisk-Source knowledge i'am not sure that my attached
patch is the correct solution for this problem. After applying my patch the
problem seems to be solved. The Asterisk is changing the SSRC & setting the
Markerbit after the transfer for the RTP-Traffic to the transfer target.




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 (0124691) addix (reporter) - 2010-07-19 03:35
 https://issues.asterisk.org/view.php?id=17007#c124691 
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Hi, i also got this issue with an unpatched Asterisk 1.4.33.1. Now we're
using twilson's patch again plus an indicate für "srcchange" for moh hold
and unhold. That fixed some RTP issues at the attendet transfer...

Sadly, the RTP to the Transfer Target (Attended transfer) still has the
same SSRC after the Transfer. How can i fix that in channel.c?

--- asterisk-1.4.33.1/channels/chan_sip.c.orig  2010-07-19
08:59:07.915495456 +0200
+++ asterisk-1.4.33.1/channels/chan_sip.c       2010-07-19
09:00:01.475353120 +0200
@@ -4158,11 +4158,13 @@
                res = -1;
                break;
        case AST_CONTROL_HOLD:
-               ast_rtp_new_source(p->rtp);
+               /* ast_rtp_new_source(p->rtp); */
+               ast_rtp_change_source(p->rtp);
                ast_moh_start(ast, data, p->mohinterpret);
                break;
        case AST_CONTROL_UNHOLD:
-               ast_rtp_new_source(p->rtp);
+               /* ast_rtp_new_source(p->rtp); */
+               ast_rtp_change_source(p->rtp);
                ast_moh_stop(ast);
                break;
        case AST_CONTROL_VIDUPDATE:     /* Request a video frame update */ 

Issue History 
Date Modified    Username       Field                    Change               
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2010-07-19 03:35 addix          Note Added: 0124691                          
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