[Users] Dealing with reinvites

[Daniel-Constantin Mierla]

Hello,

without the full grep of the packages, I am doing just assumptions.

The ACK is for first INVITE, so the the Asterisk should match it against 
the first transaction it created. For the second INVITE, it should wait 
for another ACK. This is a common race in IP world due to different path 
routing and cannot be avoided, even if you fix something on openser, the 
delay may occur on the wire.

The ACK sent for a 200OK is a separate transaction, and it is routed 
according to Route headers. You cannot change it in OpenSER/SER unless 
you make the proxy to be call stateful, to act as an end User Agent.

I am not sure what the RFC says exactly about how the end User Agent 
must act. It may be logically to drop the re-INVITE if the ACK for the 
first invite didn't arrive or create a temporary transaction and wait 
for both ACKs.

Cheers,
Daniel


On 12/30/05 17:09, Matt Schulte wrote:
> All, 
>
> 	We have had a new issue arise within openser, basically we
> connect to a SIP carrier that uses Asterisk as a proxy. Naturally, this
> carrier is hellbent on offloading the RTP traffic, thus the reinvites.
> My issue is that they appear to be sending the second invite "too fast",
> before the initial RTP even gets a chance to establish. This was working
> just fine before, and I believe they changed something. Anyway, they're
> refusing to help with this issue, so now I must "fix" it on my end.
> OpenSER is in-part, the problem.. See below:
>
> (Sorry if formatting is off, dump from tethereal ..)
>
>
>   0.000000 204.13.233.13 -> 206.80.70.47 SIP/SDP Request: INVITE
> sip:8886963856 at 206.80.70.47, with session description
>   0.001752 206.80.70.47 -> 204.13.233.13 SIP Status: 100 trying -- your
> call is important to us
>   0.002723 206.80.70.47 -> 206.80.70.54 SIP/SDP Request: INVITE
> sip:+1314xxxxxxx at sip.stl.netlogic.net, with session description
>   0.010327 206.80.70.54 -> 206.80.70.47 SIP Status: 100 Trying
>   0.292175 206.80.70.54 -> 206.80.70.47 SIP Status: 180 Ringing
>   0.292692 206.80.70.47 -> 204.13.233.13 SIP Status: 180 Ringing
>   2.016246 206.80.70.54 -> 206.80.70.47 SIP/SDP Status: 200 OK, with
> session description
>   2.016893 206.80.70.47 -> 204.13.233.13 SIP/SDP Status: 200 OK, with
> session description
>   2.050871 204.13.233.13 -> 206.80.70.47 SIP Request: ACK
> sip:+1314xxxxxxx at 206.80.70.54
>   2.051233 204.13.233.13 -> 206.80.70.47 SIP/SDP Request: INVITE
> sip:+1314xxxxxxx at 206.80.70.54  2.051827 206.80.70.47 -> 204.13.233.13
> SIP Status: 100 trying
> --<<KABOOM>>--
>   2.051956 206.80.70.47 -> 206.80.70.54 SIP/SDP Request: INVITE
> sip:+1314xxxxxxx at 206.80.70.54, 
>   2.052112 206.80.70.47 -> 206.80.70.54 SIP Request: ACK
> sip:+13142664000 at 206.80.70.54
>   2.053028 206.80.70.54 -> 206.80.70.47 SIP/SDP Status: 200 OK, with
> session description
>   2.053336 206.80.70.54 -> 206.80.70.47 SIP Status: 503 Server error
>
>
> Where you see "kaboom", is the problem. If you notice, the time between
> the first ACK and second INVITE is about 1ms .. While this shouldn't be
> a problem, you'll notice when the ACK/INVITES are getting statefully
> forwarded, they're getting sent out of order. The end point is an
> Asterisk machine, thus the 503 error.
>
> Carrier_Ast --> OpenSER --> Netlogic_Ast
>
> I am using loose_route to forward, just like everyone else, they are
> even hitting loose_route in the correct order.. Thoughts? Suggestions?
> This seems like something internal to OpenSER, I have tried butchering
> the config to force the ACK out first and it just created more problems.
> Thanks!
>
>
> 	Matt S
>
>
>
>
>
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