Anthony Messina
2011-Dec-02 17:37 UTC
[asterisk-users] CSipSimple audio issue with DAHDI/IAX2 calls
I've just connected my new Android (Motorola RAZR) phone to Asterisk
using CSipSimple and have discovered that on any call between CSipSimple
and an Asterisk DAHDI or IAX2 channel, the 'other' end of the call will
hear a rhythmic tapping as if my voice stream is being chopped up in
equal parts about every 500ms or so. I can always hear the remote party
without issue, regardless of the channel type.
The issue occurs only on connections to DAHDI channels (even those that
don't pass through the PSTN), and IAX2 connections to remote Asterisk
servers.
This issue occurs whether I am using WiFi, 3G or 4G connections on the
Android.
This does NOT occur on any SIP channels, local to my Asterisk box, or to
others.
I've investigated changing just about every setting on the Android with
no resolution. It seems like some sort of timing issue and is strange
to me that this issue is confined to DAHDI and IAX2 channels, but I'm no
expert.
I have tested using only res_timing_dadhi.so since I have the card, but
that did not help either.
Would anyone be willing to point me in the right direction for resolving
this issue? Please let me know if any more information is required.
Thanks in advance. -A
I am currently using the following on a Fedora 15 x86_64 system:
Asterisk 1.8.7.1 built by mockbuild @ x86-13.phx2.fedoraproject.org on a
x86_64 running Linux on 2011-10-17 21:42:11 UTC
]# cat /proc/dahdi/*
Span 1: WCTDM/4 "Wildcard TDM400P REV I Board 5" (MASTER)
1 WCTDM/4/0 FXOKS (In use) (EC: OSLEC - INACTIVE)
2 WCTDM/4/1 FXOKS
3 WCTDM/4/2 FXSKS (In use) (EC: OSLEC - INACTIVE)
*CLI> module show like timing
Module Description Use Count
res_timing_dahdi.so DAHDI Timing Interface 0
res_timing_pthread.so pthread Timing Interface 0
res_timing_timerfd.so Timerfd Timing Interface 1
*CLI> core show settings
PBX Core settings
-----------------
Version: 1.8.7.1
Build Options: LOADABLE_MODULES
Maximum calls: Not set
Maximum open file handles: Not set
Verbosity: 3
Debug level: 0
Maximum load average: 0.000000
Minimum free memory: 0 MB
Startup time: 10:23:07
Last reload time: 10:23:07
System: Linux/2.6.32-131.2.1.el6.x86_64 built by
mockbuild on x86_64 2011-10-17 21:42:11 UTC
Default language: en
Language prefix: Enabled
User name and group: /
Executable includes: Disabled
Transcode via SLIN: Enabled
Internal timing: Enabled
Transmit silence during rec: Disabled
Generic PLC: Enabled
--
Anthony - http://messinet.com - http://messinet.com/~amessina/gallery
8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E
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Anthony Messina
2011-Dec-28 21:03 UTC
[asterisk-users] [SOLVED] Re: CSipSimple audio issue with DAHDI/IAX2 calls
On 12/02/2011 11:37 AM, Anthony Messina wrote:> I've just connected my new Android (Motorola RAZR) phone to Asterisk > using CSipSimple and have discovered that on any call between CSipSimple > and an Asterisk DAHDI or IAX2 channel, the 'other' end of the call will > hear a rhythmic tapping as if my voice stream is being chopped up in > equal parts about every 500ms or so. I can always hear the remote party > without issue, regardless of the channel type. > > The issue occurs only on connections to DAHDI channels (even those that > don't pass through the PSTN), and IAX2 connections to remote Asterisk > servers. > > This issue occurs whether I am using WiFi, 3G or 4G connections on the > Android. > > This does NOT occur on any SIP channels, local to my Asterisk box, or to > others. > > I've investigated changing just about every setting on the Android with > no resolution. It seems like some sort of timing issue and is strange > to me that this issue is confined to DAHDI and IAX2 channels, but I'm no > expert. > > I have tested using only res_timing_dadhi.so since I have the card, but > that did not help either. > > Would anyone be willing to point me in the right direction for resolving > this issue? Please let me know if any more information is required. > Thanks in advance. -AEnabling the jitterbuffer=yes on the iax channel and setting Set(JITTERBUFFER(fixed)=default) prior to any calls to DAHDI channels seems to resolve the issue for now. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 198 bytes Desc: OpenPGP digital signature URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111228/388e4c43/attachment.pgp>