[asterisk-bugs] [Asterisk 0012247]: Choppy voicemail recording for SIP calls via TELES softswitch

[noreply at bugs.digium.com]

The following issue has been SUBMITTED. 
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http://bugs.digium.com/view.php?id=12247 
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Reported By:                fabianhoppe
Assigned To:                
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Project:                    Asterisk
Issue ID:                   12247
Category:                   Applications/app_voicemail
Reproducibility:            always
Severity:                   major
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:             03-18-2008 05:07 CDT
Last Modified:              03-18-2008 05:07 CDT
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Summary:                    Choppy voicemail recording for SIP calls via TELES
softswitch
Description: 
PROBLEM SUMMARY
We're experiencing severe audio problems with (voicemail) recordings on
our Asterisk servers for calls originating from a SIP provider using a
TELES softswitch/PSTN-gateway. 

Calls originating from a __PSTN-source__, encoded from SS7 to VOIP by the
TELES gateway and then routed via his TELES softswitch to us show strange
audio artefacts. 

Calls originating from a _VOIP source_, bypassing the providers PSTN
gateway but still being routed via the TELES softswitch don't show the
artefacts.

Calls coming in from other SIP providers (using other softswitches) are
recorded just fine. 

The problem is reproduceable and is related to the recording part as the
record application shows the some behaviour. The perceived audio quality is
in any case very good. Replacing the call to voicemail by the echo
application shows a perfect audio quality.

PROBLEM SYMPTOMS
While systems announcements (especially voicemail menus and personal
announcements) are replayed cristal clear, the recordings are often choppy
and miss the audio signal in a range of fractions of a second up to several
seconds. 

Please refer to the attached WAV files which were recorded in parallel,
one originated in the PSTN coming in via the TELES gateways and the TELES
softswitch as SIP call w/ GSM codec and one originating from a VOIP source,
coming bypassign the TELES gateway but being still routed through the TELES
softswitch.

The problem is reproducable using the ALAW codec.

OUR FINDINGS SO FAR
- RTP packet size is 20ms for all calls
- Jitter is not a problem (<5ms)
- Packet loss is not a problem (0%)
- Enabling the SIP jitterbuffer nevertheless (including the JB for app
patch as proposed by Russell Bryant) has no effect
- The network connection between the TELES softswitch and our servers is
irrelevant to the issue as it can be reproduced with an Asterisk server
installed in the same GBit-LAN as the softswitch
- The load of the system is not relevant (the test systems have a load
near 0)
- zaptel-precision is not an issue (and 99,99% on all system anyway)
- problem can be reproduced with using format_wav and format_wav_gsm
- problem persistes even if the 32KB writebuffer for the write-function is
enabled (as proposed by another bug report / feature request in mantis)

MY QUESTIONS
- Can someone analyse the attached tcpdump for any differences in the two
audio stream which might cause the audio artefacts? Are there any
differences?
- Are implementation variances for the GSM and ALAW codecs possible?
- Any ideas but chosing another SIP provider?
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Issue History 
Date Modified   Username       Field                    Change               
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03-18-08 05:07  fabianhoppe    Asterisk Version          => 1.4.18          
03-18-08 05:07  fabianhoppe    SVN Branch (only for SVN checkou => N/A          
  
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