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

[Bogdan-Andrei Iancu]

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