[OpenSIPS-Users] T.38 detection/redirect in OpenSIPS

[TCB]

@ All,

This has very exciting possibilities for me. I'm definitely going to look at
the
b2b module and test this out. I will share my findings.

regards

On Thu, Mar 18, 2010 at 10:32 AM, Bogdan-Andrei Iancu <
bogdan at voice-system.ro> wrote:

> Hi Jeff,
>
> as opensips will act as b2b, your call will be actually split in 2 calls
> (from SIP point of view) - a call C1 from GW to opensips and another one
> C2 from opensips to UAC. So at re-INVITE time, opensips b2b will hung up
> C2 and replace it with a C3 to a new destination, bridging it with C1
>
> Regards,
> Bogdan
>
> Jeff Kronlage wrote:
> > I'm confused on this as well - wouldn't you be effectively placing two
> > calls (one via a non-T38 gateway, one via a T38 gateway) to the same
> > destination?  Figuring that most T38 is going to terminate to a single
> > analog device, I would think that were this possible at a SIP level, the
> > device would already be "busy" before the second call came in as fax
> > machines don't typically drop the line very rapidly?
> >
> > Jeff
> >
> > -----Original Message-----
> > From: users-bounces at lists.opensips.org
> > [mailto:users-bounces at lists.opensips.org] On Behalf Of Bogdan-Andrei
> > Iancu
> > Sent: Wednesday, March 17, 2010 11:23 AM
> > To: OpenSIPS users mailling list
> > Subject: Re: [OpenSIPS-Users] T.38 detection/redirect in OpenSIPS
> >
> > right, that is exactly what the b2b is up to do - to be able (at
> > signalling level) to manipulate the call legs
> >
> > Regards,
> > Bogdan
> >
> > Brett Nemeroff wrote:
> >
> >> Bogdan,
> >> But at this point, you are now playing with a dialg that is already
> >> connected to an endpoint. You'd need to drop the first call to
> >> establish a new call with the reinvite. Right?
> >> -Brett
> >>
> >> On Mar 17, 2010, at 11:50 AM, Bogdan-Andrei Iancu
> >>
> > <bogdan at voice-system.ro
> >
> >>  > wrote:
> >>
> >>
> >>
> >>> Hi Brett,
> >>>
> >>> Brett Nemeroff wrote:
> >>>
> >>>
> >>>> I don't think there is any way to do this without an RTP capable
> >>>> device in the mix.
> >>>>
> >>>>
> >>> you do not need to look into RTP as the FAX is advertised in the
> >>> re-INVITE (in SDP) - so you can detect it from opensips script by
> >>> inspecting the SDP of reINVITES
> >>>
> >>>
> >>>> What you may be able to do is have asterisk detect that it's a fax,
> >>>> then reject it if it is.. I don't know if you can do all that
> >>>>
> > without
> >
> >>>> answering the call.
> >>>>
> >>>>
> >>> no, you cannnot, as first the call is established (from sip point of
> >>> view) as a simple audio call and after that re-negotiated (via
> >>> re-INVITE) for FAX
> >>>
> >>>
> >>>> Then you can forward it back to the proxy if it is a fax with maybe
> >>>>
> > a
> >
> >>>> prefix.
> >>>>
> >>>> A lot of assumptions in there. Would like to hear if you find
> >>>> something that works. Not sure if you can SIP Spiral yet in asterisk
> >>>> anyway. ;)
> >>>>
> >>>>
> >>> I do not see the need of Asterisk - maybe with some changes, the b2b
> >>> module will be able to handle this - see my prev email.
> >>>
> >>> Regards,
> >>> Bogdan
> >>>
> >>>
> >>>
> >>>> -Brett
> >>>>
> >>>>
> >>>> On Wed, Mar 17, 2010 at 10:51 AM, David J. <david at styleflare.com
> >>>> <mailto:david at styleflare.com>> wrote:
> >>>>
> >>>>    Matt,
> >>>>
> >>>>    I am for sure probably wrong, but I think you would need
> >>>> Asterisk or
> >>>>    Variant to Determine that it is a Fax Call,
> >>>>    I dont think UAC's send T38 information without negotiating with
> >>>> the
> >>>>    other side who request that it is capable, then it brings you to
> >>>>    Jeff's
> >>>>    answer.
> >>>>
> >>>>    See above.
> >>>>
> >>>>
> >>>>    Matthew S. Crocker wrote:
> >>>>
> >>>>
> >>>>> Can OpenSIPS make routing decisions based on the SDP information
> >>>>>
> >>>>>
> >>>>    in an INVITE?
> >>>>
> >>>>
> >>>>> Lets say I have the following config
> >>>>>
> >>>>> PSTN -> t.38 Gateway -> OpenSIPS ->  UserAgent
> >>>>>
> >>>>> I have a TN from the PSTN routed to the UserAgent,  I'd like to
> >>>>>
> >>>>>
> >>>>    provide a service so the user can use the TN for both voice &
> >>>> faxing.
> >>>>
> >>>>
> >>>>> Voice call goes through normally (g.711 g.729 codec)
> >>>>>
> >>>>> Fax call starts off as a normal voice call (INVITE, 180, 183,
> >>>>>
> >>>>>
> >>>>    200).  Once the call is answered the originating end (PSTN)
> >>>>
> > starts
> >
> >>>>    sending fax tones. The Gateway hears the fax tones and attempts
> >>>>
> > to
> >
> >>>>    RE-INVITE with T.38 in the SDP.  I'd like OpenSIPS to see the
> >>>>
> > T.38
> >
> >>>>    capability in the SDP and redirect the call to a fax->e-mail
> >>>>    gateway.  So,  the 2nd INVITE comes in, OpenSIPS sends the INVITE
> >>>>    to the fax gateway and a BYE to the user.  The fax gateway does a
> >>>>    200 and negotiates T.38 with the PSTN gateway.
> >>>>
> >>>>
> >>>>> I know I can route the call through Asterisk and have it do a
> >>>>>
> >>>>>
> >>>>    quiet answer and listen for the modem sounds.  I'd like to avoid
> >>>>    using Asterisk for all RTP traffic and only use it for the fax
> >>>>    gateway traffic (i.e. once it has been determined to be a fax
> >>>>    Asterisk steps in and handled the T38 -> E-mail)
> >>>>
> >>>>
> >>>>> -Matt
> >>>>>
> >>>>>
> >>>>>
>
>
> --
> Bogdan-Andrei Iancu
> www.voice-system.ro
>
>
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>



-- 
TC
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