[Asterisk-Dev] Recommendations for SIP PBX/SP interoperability draft

[John Todd]

http://wiki.sipforum.org/images/a/a9/Sf-draft-twg-IP_PBX_SP_Interop-sibley-v3.pdf

Those of you using Asterisk in commercial environments, specifically 
those of you re-packaging Asterisk as a "solution" or as a "service 
provider gateway" should read the above document.  It describes the 
work to date of the SIPConnect folks on how to make PBX/SIP platforms 
talk to each other on the "public" Internet.  The document as it 
stands currently is really designed as a guideline for how service 
providers can use SIP as a trunk to let customer devices talk to 
their networks, but I expect it will be the guideline that gets used 
when more entities are using SIP to communicate between each other 
without a service provider.  <cough>ENUM<cough, wheeze>

There is useful data in here for those programming Asterisk for 
future compliance:

   - mappings of SS7 to SIP error codes
   - media negotiation
   - VAD, echo cancellation, and codec options

Frequently referenced in this document and notably absent from 
Asterisk is the use of TCP and TLS for SIP, which I fear is one of 
the major shortcomings of the VoIP aspects of Asterisk currently. 
I'll donate to a coder, but I don't have the code...  There are 
plenty of other RFCs listed that would be well-suited for Asterisk 
programmers to understand, either in the code itself, or in the 
dialplan.

JT