Yes, I realise Asterisk is not a media relay, but I don’t think the OP is using a pure media relay, otherwise the REFER message would not be sent to it, would it?

Thank you for the hash table suggestion by the way.

Charles

From: sr-users-bounces@lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Reda Aouad
Sent: 26 April 2012 15:15
To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - UsersMailing List
Subject: Re: [SR-Users] dispatcher and call transfer

Asterisk is not really a media relay. Asterisk establishes 2 legs for each call, and I'm not sure what happens in this case.

An improvement you can make is to use a hash table (in-memory) to store the information you need (call-id, from, to) then lookup in
the htable for existing calls for same users. That should relieve your database from a query on every invite and increase performance if you have a large number of calls.

Reda

On Thu, Apr 26, 2012 at 15:52, Charles Chance <charles.chance@sipcentric.com> wrote:

Hi Reda,

Sorry, I should have been more specific – I am referring to instances where the media server is for example Asterisk. If first call goes through Asterisk 1, second call goes through Asterisk 2, when Asterisk 2 receives the REFER it does not know of initial call on Asterisk 1. The only way we have found for it to work is to ensure the second call is dispatched to the same Asterisk box as the first.

I would be pleased to hear of an alternative method.

Regards,

Charles

From: sr-users-bounces@lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Reda Aouad
Sent: 26 April 2012 14:34

To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - UsersMailing List
Subject: Re: [SR-Users] dispatcher and call transfer

Hi,

@Carsten
Dispatcher algorithm 0 based on call-id should do it in your case of re-invite within dialog with same call-id.

@Charles
In the case of attended transfers, shouldn't both media servers be relaying media between them? I didn't understand why your are obliged to dispatch to the same media server since they are 2 different calls with different call-ids.

Reda

On Thu, Apr 26, 2012 at 14:30, Charles Chance <charles.chance@sipcentric.com> wrote:

Hi,

Actually, this won't help for attended transfers where another call is
initiated first then the two are joined together by REFER. In this case, the
second INVITE must be routed to the same media server as the existing call
for the transfer to work.

What we do is store the dialogs in DB, then when a new call comes in, prior
to doing ds_select_dst we query DB for existing call involving same user. If
we find one, we simply replace destination host with that from the contact
(to/from depending on direction of call).

It may not be the most elegant way but it works for us :)

Charles

-----Original Message-----
From: sr-users-bounces@lists.sip-router.org
[mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Carsten Bock
Sent: 26 April 2012 13:25
To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
UsersMailing List
Subject: Re: [SR-Users] dispatcher and call transfer

Hi,

if you look at the docs of the dispatcher module, you'll find this:

alg - the algorithm used to select the destination address. The
parameter can be an integer or a variable holding an interger.
- “0” - hash over callid
(http://kamailio.org/docs/modules/devel/modules_k/dispatcher.html#id2498492)

But probably you should look into record/loose_route for your setup.
Since REFER is normally an in-dialog request (belongs to another
voice-session), it should take the same route as the initial INVITE.
This is normally achieved by the record/loose-route mechanisms
described in RFC3261.
In the example config of Kamailio you find an configuration example (below).

It is not a bug in the dispatcher module, it's how you use it.

So long,
Carsten

455 # handle requests within SIP dialogs
456 route(WITHINDLG);

[...]

473 # record routing for dialog forming requests (in case
they are routed)
474 # - remove preloaded route headers
475 remove_hf("Route");
476 if (is_method("INVITE|SUBSCRIBE"))
477 record_route();

and the relevent parts in the "WITHINDLG" route:

566 # Handle requests within SIP dialogs
567 route[WITHINDLG] {
568 if (has_totag()) {
569 # sequential request withing a dialog should
570 # take the path determined by record-routing
571 if (loose_route()) {
[...]
580 route(RELAY);
581 } else {
[...]
587 if ( is_method("ACK") ) {
588 if ( t_check_trans() ) {
589 # no loose-route, but stateful
ACK;
590 # must be an ACK after a 487
591 # or e.g. 404 from upstream
server
592 t_relay();
593 exit;
594 } else {
595 # ACK without matching
transaction ... ignore and discard
596 exit;
597 }
598 }
599 sl_send_reply("404","Not here");
600 }
601 exit;
602 }
603 }

2012/4/26 Asgaroth <00asgaroth00@gmail.com>:
> Hi All,
>
> Currently we are running kamailio in a loadbalanced fashion whereby calls
> come in via the loadbalancers and distribute calls accross 2 media
servers.
> We have come accross and issue whereby call transfers may be distributed
> accross two media servers and when the REFER message comes along to
transfer
> the call, in some cases (if we're lucky) the message arrives at the wrong
> media server (transaction leg doesnt exist).
>
> Some googling later and it appears that dispatcher doesnt play nice when
it
> comes to this scenario. Some suggestions popped up in my previous searches
> saying that a potential work around is to use the dialog module to check
if
> a call is eastablished and then to send all calls to the same media server
> based on the dialog already being established.
>
> I'd appreciate some input from the guru's out there that have come accross
> this same issue and, if possible, some suggestions on how to work around
the
> problem, does the dispatcher module have a hashing algorithm that can be
> suited for this particular scenario?
>
> Thanks in advance for any tips or sugestions.
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>

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