[asterisk-dev] Provider requires Delay between OK and ReINVITE

[Olle E. Johansson]

22 aug 2012 kl. 16:52 skrev John R Covert:

> Help, please.
> 
> I have a client using a provider which has just installed a new
> SIP trunk into our Asterisk installation from a different platform.
> All works fine on their "Sonus" platform.  However, on their "Fusion"
> platform, they drop a call when they receive a ReINVITE.
> 
> 09:48:58.495468 IP PROVIDER > ASTERISK: SIP, length: 1001
> INVITE sip:exten at asterisk:5060 SIP/2.0
> 
> 09:48:58.501365 IP ASTERISK > PROVIDER: SIP, length: 503
> SIP/2.0 100 Trying
> 
> 09:48:58.886713 IP ASTERISK > PROVIDER: SIP, length: 519
> SIP/2.0 180 Ringing
> 
> 09:49:12.794858 IP ASTERISK > PROVIDER: SIP, length: 815
> SIP/2.0 200 OK
> 
> 09:49:12.795835 IP PROVIDER > ASTERISK: SIP, length: 397
> ACK sip:exten at asterisk:5060 SIP/2.0
> 
> 09:49:12.801258 IP ASTERISK > PROVIDER: SIP, length: 835
> INVITE sip:2167129661 at 208.93.226.215:5060 SIP/2.0
> 
> 09:49:12.801595 IP PROVIDER > ASTERISK: SIP, length: 336
> SIP/2.0 100 Trying
> 
> 09:49:12.845627 IP PROVIDER > ASTERISK: SIP, length: 508
> SIP/2.0 491 Request Pending
> 
> 09:49:12.848064 IP PROVIDER > ASTERISK: SIP, length: 614
> BYE sip:exten at asterisk:5060 SIP/2.0
> 
> They have asked me to provide a delay between the "OK" sent
> by Asterisk and the ReINVITE.
> 
> Can this be done in the existing code?  If I were to develop a
> change to SIP to allow this to be configured on a per-peer
> basis or some other way (please suggest) would it be accepted
> into the code base.
We currently have no delay between the original call setup
and the reinvite process. It could of course be made to a configurable
option like you say and I see no reason why a patch that does this would
not be accepted (unless the code is really bad… :-) )

This option could be like this:

directmediadelay = 10    ; Delay from bridge setup to media optimization attempt
                                            ; default=0 s


You can of course also disable the re-invites (direct media) in order
to be able to use the new SIP trunk, but it sounds from your mail
like this is something you want to happen.

/O
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