[asterisk-bugs] [Asterisk 0012378]: lost packets when using Polycom 550 and ulaw codec to outbound sip also using ulaw, if change polycom to alaw no lost packets

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=12378 
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Reported By:                tallen8840
Assigned To:                
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Project:                    Asterisk
Issue ID:                   12378
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.18 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             04-07-2008 15:38 CDT
Last Modified:              06-05-2008 19:11 CDT
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Summary:                    lost packets when using Polycom 550 and ulaw codec
to outbound sip also using ulaw, if change polycom to alaw no lost packets
Description: 
rtcp stats showing lots of lost packets and bad voice quality and Polycom
stats show same number of lost packets when placing a call from A Polycom
550 to and outbound (myphonecompany.com) SIP "trunk" using the ulaw codec
at both ends. No problems when call terminates at Asterisk (my
AutoAttendant) or calling another SIP extension Cisco ATA186 etc) or
calling from another SIP extension (Cisco ATA) to outbound trunk with ulaw
at both ends. So after checking lots of other stuff I changed the Polycom
sip entry to alaw, and now it's all better. Please feel free to contact me
you can get my phone number from www.integsys.biz. I will provide xml
(Polycom), logs whatever from the phone, Asterisk, etc.

Debian Testing quad xenon 700
Polycom 550 SIP 2.2.2.0084



Asterisk 1.4.18.1~dfsg-1 built by pbuilder @ grnetbox on a x86_64 running
Linux on 2008-03-18 22:54:26 UTC

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---------------------------------------------------------------------- 
 jolan - 06-05-08 19:11  
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Can you try this?

This seems to workaround the issue for me.

Index: rtp.c
===================================================================
--- rtp.c	(revision 120859)
+++ rtp.c	(working copy)
@@ -2663,8 +2663,10 @@
     if (rtp->lastts > rtp->lastdigitts)
         rtp->lastdigitts = rtp->lastts;
 
+	/* XXX polycoms can't cope with high timestamps on the initial packet
     if (ast_test_flag(f, AST_FRFLAG_HAS_TIMING_INFO))
         rtp->lastts = f->ts * 8;
+	*/ 

Issue History 
Date Modified   Username       Field                    Change               
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06-05-08 19:11  jolan          Note Added: 0087876                          
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