I'm not sure what the standard says, but AFAIK the dialog is confirmed with 200 Ok from the first INVITE. Thus, Asterisk should accept the new reINVITE although the ACK was not received yet. (IMO, receiving an reINVITE imply that the previous 200 Ok was ACKnowledged).

IMO Asterisk should accept the reINVITE (just like when the ACK was received, thus stopping retransmissions). If then the ACK is received, the CSeq does not match, thus it should be ignored.

Which Asterisk version do you use? 1.2.1? Try to discuss the problem on the devel mailing list, and create a bug report on bugs.digium.com with suggested proposals how to fix the problem.

regards
klaus

Matt Schulte wrote:
I see what you mean, and was worried that would be the answer .. :-)


The ACK is for first INVITE, so the the Asterisk should match it

against the first transaction it created.

Since the cseq's are different, Asterisk flips out, since it expects the
102 cseq before the 103 .. This is normal behavior, right?


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.

Well, the cats out of the bag, this carrier doesn't want to disable
reinvites.. I think they have altered the Asterisk code to send a
reinvite ASAP as opposed to 'waiting' for the RTP stream to setup. We
have never ever had this problem before.. For those of you wondering,
the carrier is TXLink..

Any other suggestions? Hehe. Thanks much.

        Matt

-----Original Message-----
From: Daniel-Constantin Mierla [mailto:[EMAIL PROTECTED] Sent: Saturday, December 31, 2005 6:55 AM
To: Matt Schulte
Cc: [email protected]
Subject: Re: [Users] Dealing with reinvites

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:[EMAIL PROTECTED], 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:[EMAIL PROTECTED], 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:[EMAIL PROTECTED]
 2.051233 204.13.233.13 -> 206.80.70.47 SIP/SDP Request: INVITE
sip:[EMAIL PROTECTED] 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:[EMAIL PROTECTED],
 2.052112 206.80.70.47 -> 206.80.70.54 SIP Request: ACK
sip:[EMAIL PROTECTED]
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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