[OpenSIPS-Users] B2BUA REFER scenario

[Tony Ward]

Thanks for explaining this, Bogdan.

I'm curious, though - would things work better if the late-SDP
negotiation were done the other way around i.e., since call3 is a new
call it could take longer for setup to occur, and during this time call1
times out.   If we were to send the initial INVITE to call3, get the
OK+SDP from call3, then invite+SDP  call1, perhaps it would respond more
quickly since the call is already set up, and when it does we can send
ACK+SDP to call3 to complete the transfer.   This may also provide the
ability in the future to avoid disconnecting call1 leg to the IVR so
that it can be retained in the event call3 fails.

Just a thought...
Tony Ward

-----Original Message-----
From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] 
Sent: Thursday, March 06, 2014 12:13 PM
To: OpenSIPS users mailling list; Tony Ward
Subject: Re: [OpenSIPS-Users] B2BUA REFER scenario

Hello Tony,

You may consider this a side effect of how the OpenSIPS B2BUA works. As
we want to be transparent from media point of view, OpenSIPS does not
get involved into SDP negotiation -> it is transparent, passing the SDPs
between the end points. This is mainly based on the late-SDP negotiation
- and combining this with normal SDP negation may lead into ACK delays.

In this particular case the ACK for Call1 is delaied because the B2BU is
waiting for the 200 OK on Call3 - because the SDP from the 200 OK in
Call3 must be pushed in the ACK for Call1 (to allow direct SDP
negotiation between the end points involved in call1 and call3). So the
ACK for call1 will be delayed until Call3 gets answered - there is no
work around this at this point. In most of the cases the UA are a bit
flexible in waiting for ACK, like at least performing 200OK
retransmission (as per RFC3261). I would say your PSTN device is a bit
picky ;) when comes to firing the BYE in 6 sec (no retransmission, short
period of time).

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 05.03.2014 20:06, Tony Ward wrote:
> Hello,
> I currently have this configuration:
> PSTN <====> SIP/ALG Router <====> OpenSips 1.10 <====> IVR
> OpenSips has a single IP on the private network.
>
> I have configured opensips using top hiding in the dialog module and
it
> works fine for calls to ptsn  and calls from pstn.
> I have also configured opensips using B2BUA top hiding and it also
works
> fine for calls to ptsn  and calls from pstn.
>
> Now I want to test B2BUA REFER scenario (where calls from PSTN are
> answered by IVR, then IVR does a REFER to another PSTN number).
>
> When the IVR sends REFER the call is dropped after .6 seconds.  The
flow
> that I've seen in the trace is below:
> PSTN			     opensips			IVR	
> 	invite+SDP (Call1) ---->  |	
> 	<----- Trying (Call1)	|
> 			    	|	invite+SDP (Call2)----->
> 			    	|	<----- OK+SDP (Call2)
> 	<----- OK+SDP (Call1)   	|
> 	Ack (Call1) ---->    	|
> 			    	| 	ACK (Call2) 	----->
>
> 		<Ivr dialog take place here>
>
> 				|	<----- REFER (Call2)
> 	<----- Invite (Call1)	|
> 			    	|	Accepted (Call2) ----->

> 			    	|	BYE (Call2) 	----->
> 	Trying (Call1) ---->           |
> 	OK+SDP (Call1) ---->       |
> 	<----- Invite+SDP(Call3)|
> 			    	|	<----- OK (Call2)
> 	Trying (Call3) ----->  	|
> 	OK+SDP (Call1) ----->  	|
> 	OK+SDP (Call1) ----->  	|
>
> 		<0.6 seconds elapse here>
>
> 	Bye (Call1) ----->	|
> 	<----- OK (Call1)   	|
> 	<----- Cancel (Call3)   	|
> 	OK (Call3) ----->	|
> 	Req Termd (Call3) ----->|
> 	<----- Ack (Call3)   	|
>
> It looks as though the PSTN times out waiting for an ACK after sending
> OK+SDP(Call1) a couple times and then waiting .6 seconds.
> The question is - what should the flow look like?  According to this
> post:
http://lists.opensips.org/pipermail/users/2012-April/021352.html,
>
> things appear to be working as expected up to the point where we
receive
> Trying (Call3).  Should I be seeing the OK+SDP from call 3 next?
> I'd like to troubleshoot further but I'm not sure where to look.
>
> Thanks!
> Tony
>
>
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