Does anybody recall a packet per second limit on ACS BXB services? Would it be any different than the NetEffect services?
On Fri, Jul 10, 2009 at 7:56 PM, Shane Spencer<shane@bogomip.com> wrote:
> OK. Here's what I've got so far.
>
> 1.) I removed all QoS from both sides and removed all routes except for
> those between sites, turned off IP forwarding on all boxes. Asterisk is
> being run on the gateway devices at Fairbanks and Wasilla, Anchorage has an
> Asterisk server with an external IP routed to over gigabit.
>
> 2.) Tested between Wasilla and Anchorage using iperf. no UDP packets had
> bad checksums. rate limited the UDP test to half of the available
> bandwidth.
>
> 3.) Tested between Wasilla and Anchorage using IAX2/Speex and an echo test.
> ~30% of the IAX/Codec packets had a bad checksum in the UDP header - call
> quality was good.
>
> 4.) Tested between Wasilla and Anchorage using IAX2/Speex over OpenVPN UDP
> tunnel mode. 0 of the UDP packets over the tunnel had a bad UDP checksum.
> ~30% of the OpenVPN UDP packets had a bad checksum in the UDP header - call
> quality was good.
>
> I'm trying to duplicate the problem I have been having for months at the
> moment, any minute now.
>
> *UPDATE* While trying to diagnose the issue, I turned off the RX/TX
> checksumming and TSO on my e1000 nics on all servers related to the calling.
> No UDP checksum issues. Sigh.. I'm starting to think that's just plain
> not important anymore :)
>
> - Shane
>
> On Fri, Jul 10, 2009 at 6:12 PM, Shane R. Spencer<shane@bogomip.com> wrote:
>>
>> 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 Wed Jul 15 18:37:32 2009
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