[asterisk-dev] Pimp My SIP merge

[Mark Michelson]

Hi!

Those of you who watch the commits list have probably seen that the 
pimp_my_sip branch[1] has been merged to trunk. The reason for this is 
that, with the exception of an API for handling incoming PUBLISH 
requests, the API for new SIP work has reached a stable point. There may 
still be forthcoming changes, but they will not be major.

So does this mean that SIP development for Asterisk 12 is complete? Not 
by a long shot!

For those of you brave enough to give what's in Asterisk trunk a whirl, 
here's a brief list of what you can do:

Basic calls (inbound and outbound)
     * Audio and video support
     * DTMF support for RFC 4733, inband, and INFO
     * Caller ID and limited Connected Line support
     * Session timers
     * PRACK
     * RFC 3326 (Reason header) supportAuthentication (inbound and outbound)
     * Direct media
Registration (inbound and outbound)
Call forwarding
Sending OPTIONs outbound
NAT traversal (including ICE support)
MWI (Just NOTIFY support, no SUBSCRIBE support)
SIP debugging
Configuration for the following items:
     * Endpoints ("peers" in chan_sip terminology)
     * Addresses of record and their contacts
     * Domains
     * Authentication
     * Transports (to include support for multiple transports)

Here's a brief list of items that are currently in development and/or up 
for review:

MESSAGE support (both in-call and out-of-call)
A media negotiation dialplan function to explicitly set codecs on 
outbound calls
SDES SRTP support
Diversion header support

Other upcoming tasks can be found on the SIP project page's JIRA issues 
section [2].

Documentation for how to configure the new SIP work is slim for now. If 
you have questions or would like to improve documentation, please feel 
free to speak up. Currently, Brad Latus has a review up adding XML 
documentation for configuration items [3]. It's a good first step 
towards making the new work more user-friendly.

This merge is a milestone, of sorts, mostly due to the API stability. 
Developers interested in adding new features should continue working 
either in the pimp_my_sip SVN branch or in a branch based off of 
pimp_my_sip. We're not sure yet when the next batch of code will make it 
into trunk, but the next batch will in all likelihood be much smaller.

Thanks for the support,
Mark Michelson

[1] http://svn.digium.com/svn/asterisk/team/group/pimp_my_sip
[2] 
https://wiki.asterisk.org/wiki/display/AST/New+SIP+channel+driver#NewSIPchanneldriver-ProjectPlanning
[3] https://reviewboard.asterisk.org/r/2471/