[asterisk-dev] SIP + T.38 == 'fax' extension?

[Steve Underwood]

Kevin P. Fleming wrote:

>When a SIP call receives a reinvite to put it into T.38 mode, maybe we should send the channel to the 'fax' extension as chan_zap does with faxdetect=yes... Thoughts?
>
>  
>
Where would the fax extension be, though? Currently the code that has 
gone into * only deals with simple passthrough cases.

I am having problems with the current way the UDPTL code works in *. I 
implemented it in a way that should be just fine by the specs. Now it is 
meeting the real world. :-(  Some boxes, especially Cisco ones, are 
causing grief. Although the reinvite/response sequence specifies a new 
port for the UDPTL traffic on the Asterisk box, some boxes continue 
sending to the original RTP port. Cisco, in particular, are not ignoring 
the new port. If it is set to zero, they don't switch from RTP to UDPTL 
at all. However, when the specified port is non-zero, they switch into 
UDPTL mode, but use the original RTP port number.

This issue probably goes beyond simple tolerance of non-compliant boxes. 
Re-using the original port is also helping with firewall/NAT issues. 
Since the path is already opened up when UDPTL commences, everything 
works smoothly when the very first UDPTL message comes from the outside.

I have been experimenting with a merged UDPTL/RTP arrangement. The UDPTL 
and RTP can then be processed properly in all the cases I have tried so far.

Regards,
Steve