[asterisk-users] SIP peer / Maximum retries exceeded on transmission

[0xception]

I can confirm the same error message... i haven't done nearly the amount of
debuggin you have but it's the exact same error message i receive when i use
a software based SIP phone connecting to another internal software SIP
phone... some times it's twinkle to xlite some times xlite to xlite and some
times twinkle to twinkle ...

That's about all that i can confirm :) hope you get some help cuz i was also
looking for some info on what the issue was.

On 5/8/07, chris at cgb1911.mine.nu <chris at cgb1911.mine.nu> wrote:
>
> (repost - can anyone confirm whether they've seen this before, or have
> any tipes in debugging it?)
>
> Hi Everyone,
>
> I was hoping someone might know why I am experiencing a problem with
> Asterisk logging the event:
>
> [May  3 12:07:41] WARNING[30371] chan_sip.c: Maximum retries exceeded on
> transmission 03f007af2b15cd0b54b0c368265d97be at sip.externalprovider.com for
> seqno 669371069 (Critical Response)
>
> This is happening after:
>   - call is setup, 2 way audio
>   - call can function correctly for up to 5 minutes, with the external
>     provider re-inviting every 1 minute
>
> When the problem happens
>   - external peer re-invites asterisk
>   - asterisk sends 200 OK
>   - external peer sends ACK
>   - asterisk retransmits 200 OK
>   - external peer sends ack
>   - ..
>   - asterisk retransmits 200 OK (Retransmitting #6)
>   - external peer sends ack
>   - Asterisk logs the above message about maximum retries exceeded,
>     and sends BYE to the inside SIP UA.
>
>
> The network configuration is as follows:
>   phone <--> alternative SIP server <--> Asterisk <-NAT-> External peer
>
> The alternative SIP server is not a B2BUA, just SIP proxy.  Now,
> sometimes a call can work without any problems, but not as often as
> when the above symptoms are experienced.
>
> The references I've found online about this type of problem suggest
> NAT as being the culprit, but in this case, Asterisk is logging it's
> reception of the ACK but deciding to ignore it and retransmit the
> 200 OK anyhow.  I'm guessing in other cases people suspect is' NAT
> because they believe SIP isn't getting back trhough after a period of
> time.
>
> I was using 1.4.2, but found this changelog today for 1.4.3:
>
> ftp://ftp.digium.com/pub/asterisk/releases/ChangeLog-1.4.3
>
> 2006-09-30 16:12 +0000 [r44068-44078] Paul Cadach <
> paul at odt.east.telecom.kz>
>   * channels/chan_sip.c: Found some buggy SIP clients (phones Planet
>     VIP-153T firmware 1.0, Linksys PAP2 firmware 3.1.9(LSc)) which
>     sends ACK not on OK message only (when remote party answers) but
>     on RINGING message too, so when we send 200 OK message, we get
>     unidentified ACK message (because INVITE acknowledged on RINGING
>     message already), so 200 OK retransmits within its retransmission
>     interval then call gets dropped. If someone else knows how to
>     provide workaround for such cases, please, fix it in correct way.
>     Thanks to ssh from #asteriskru for provide access to his box to
>     study and fix this case.
>
> I've upgraded to 1.4.4 but the problem still persists.  The above
> changelog doesn't sound exactly like what I"m experiencing but maybe
> it's related.
>
> Attached is my sip.conf, extensions.conf, and (debug = 10) logs for
> one example.  I don't know what else might be needed to help anyone
> assist me in this problem - let me know if I missed something.
>
> It *feels* like an Asterisk bug but maybe a SIP expert can spot the
> problem in signalling/RFC conformance..
>
> Thanks in advance,
>
> Chris Bennett
>
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