Hey, Royce -
Royce Williams wrote:
> Hey, Shane -
>
> Shane R. Spencer wrote, on 7/10/2009 2:33 PM:
>> My last resort will be IAX2 + Speex.. but for now I ask the jury.. What t=
>> he heck is going on with my ACS services. I have a BXB in Fairbanks
>> and Anchorage and ACS DSL in Wasilla. Fairbanks and Wasilla have several=
>> VoIP phones each. Usually only one g.729a call at a time.
>> Generally only 20msec of audio is sent per packet per channel. Even with=
>> no site transfering data or browsing the web, I get calls that seem to
>> have packet out of order or queueud packets. Wherein 20msec is sent per =
>> packet.. there seems to be a short burst of packets instead of a fluid
>> stream. Audio wise it seems like the playback is constantly trying to ca=
>> tch up with either buffered or lost packets. I need to do lower level
>> diagnosticts.
>
> Speaking as a geek, and not as an ACS droid (yet, anyway), a couple of
> questions:
>
> - Sounds like it was working better before; when did it go south?
Approximately 4 weeks ago it went further south than it was before. It started with intra-ACS connections then we started noticing poor call
quality with our SIP peers in the lower 48. I am setting up some testing in the lower 48 so I can check out RTP checksums.
> - Have you sniffed the traffic to look for actual retransmits or other
> 'smoking gun' evidence of drops?
No retransmits since it's all RTP, however I see a LOT of RTP checksum issues. I'm tempted to turn off RTP checksumming in iax.conf and watch
the errors fly with codec_speex.so. I'm not at that point yet.. still trying to wrangle a goat up for the obligatory sacrifice.
Shane
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Received on Fri Jul 10 18:13:22 2009
This archive was generated by hypermail 2.1.8 : Fri Jul 10 2009 - 18:13:23 AKDT