[VoiceOps] T.38 re-invites

[Mark R Lindsey]

It's true: you can go against the conventional "best practice" that everyone else is following. 

But you cannot expect that everybody who looked at the IETF draft for their implementation planning is really implementing all of the possible combinations of features and permutations of operations. Especially when DSPs are on the line, vendors are very parsimonious about additional features. Some would consider the ability to accept or generate a T38 re-invite in either calling or called states to be an additional feature.

So if you are going to deviate from the BCPs, you should expect to have some interop fun. 

I have given up the dream that every developer will implement every RFC in its fullest generality.

On Jan 10, 2012, at 1:15, Ryan Delgrosso <ryandelgrosso at gmail.com> wrote:

> Best practice and real-world are often very different things.
> 
> I have seen numerous scenarios where a softswitch acting as a forking proxy will use a dummy SDP in its outgoing invite (Metaswitch and Sylantro both come to mind here), and immediately will re-invite when the far end has answered.
> 
> I have also seen numerous scenarios where fax servers will send the initial invite with G711 and then instantly re-invite to T.38 upon answer.
> 
> Dont stand on convention or BCP here, the re-invite often comes from the calling party, and sometimes a few times.
> 
> 
> 
> On 01/04/2012 06:46 AM, Mark R Lindsey wrote:
>> The IETF informational draft: "SIP Support for Real-time Fax: Call Flow Examples And Best Current Practices" shows the re-INVITEs from the called party. This is still conventional best practice.
>> 
>>    http://tools.ietf.org/html/draft-ietf-sipping-realtimefax-01
>> 
>> 
>> Mike Coffee's talk at SIPNOC 2011 discussed the expectations, showed how things can fall apart:
>> 
>>    http://www.sipforum.org/component/option,com_docman/task,cat_view/gid,115/Itemid,261/
>> 
>> 
>> -- Mark R Lindsey mark at ecg.co +1-229-316-0013  http://ecg.co/lindsey
>> 
>> 
>> 
>> 
>> On Jan 4, 2012, at 04:35 , Nathan Anderson wrote:
>> 
>>> This might not be the most appropriate forum for such a question, but given that implementation bugs in either NOC gear or CPE gear can cause headaches for network operators, I offer this here in the hope that perhaps a fellow VoiceOp here may have run into this before and knows the answer...
>>> 
>>> It is my understanding that with SIP, *either* party to a call can send another INVITE (a so-called "re-invite") at any time during an active session; this could be used to change the endpoint of a call or redirect it elsewhere, and is also commonly used to renegotiate the media in use (audio, video, image, and associated codecs) without dropping the call even while the endpoints remain the same.
>>> 
>>> Does this hold true, though, for re-invites whose purpose is to switch from an RTP audio stream to a T.38 UDPTL stream?  Can *either* peer send that INVITE?  Or is there a rule -- spoken or unspoken -- that states that a T.38 re-invite can only be initiated by the *receiver* of the fax, and not the transmitter?
>>> 
>>> I have some ATAs that appear to send out a T.38 re-invite upon detection of a fax tone regardless of whether the ATA placed the call or answered the call, and if I am understanding what I'm seeing in my packet captures, I think the fact that these ATAs do this even when they are going to play the role of the T.38 transmitter is confusing the heck out of my (Asterisk) SBC and the termination provider.  But I can't find any concrete evidence to support the idea that T.38 re-invites should only be initiated by the receiver.
>>> 
>>> -- 
>>> Nathan Anderson
>>> First Step Internet, LLC
>>> nathana at fsr.com
>>> _______________________________________________
>>> VoiceOps mailing list
>>> VoiceOps at voiceops.org
>>> https://puck.nether.net/mailman/listinfo/voiceops
>> 
>> _______________________________________________
>> VoiceOps mailing list
>> VoiceOps at voiceops.org
>> https://puck.nether.net/mailman/listinfo/voiceops
> _______________________________________________
> VoiceOps mailing list
> VoiceOps at voiceops.org
> https://puck.nether.net/mailman/listinfo/voiceops