[asterisk-dev] Proposal for T.38 transparent gateway design in Asterisk

[Steve Underwood]

On 04/30/2010 04:57 AM, Andreas Sikkema wrote:
>>> Also
>>>    SIP-ATA 1<---SIP+T.38--->Asterisk<---SIP+G711---->  SIP-ATA 2
>>> is a valid scenario and should be supported. In this case Asterisk
>>> should reINVITE to G711 if the T.38 reINVITE was rejected by SIP-ATA 2.
>>>        
>> I would say this should *not* be supported; if someone has a FAX machine
>> attached to a SIP ATA, and that ATA does not support T.38, then they
>> should get one that does. Even if we did support this scenario (assuming
>> SIP-ATA1's FAX machine is the caller), the call is likely to not switch
>> to T.38 at all since there is no gateway that will detect the FAX
>> preamble on a TDM channel. If you configured Asterisk to do that, it
>> would detect the preamble from SIP-ATA2 and send a T.38 re-INVITE to
>> SIP-ATA2, not SIP-ATA2.
>>      
> There's a sizable amount of SIP ATA's in the wild that offer the ability to configure an analogue port to "fax" port and therefore start each call from that port on the PSTN side straight with a T.38 offer in the SDP, no re-invite's needed.
>    
Try setting those ATAs to the "always a FAX" mode and you will find most 
still re-invite to T.38. If they didn't they would be unable to try the 
fallback position of FAXing by G.711 - an unreliable option, but still 
worth a try when T.38 is not an available option. It is rare for 
something to start in T.38 mode. It just doesn't work for a lot of 
devices supporting T.38. They *require* a re-invite to T.38. Even 
terminals incapable of anything but T.38 operation, like t38modem, start 
in audio mode for this compatibility reason.

Steve