[asterisk-dev] AstriDevCon 2015 SemiAnnual Status Check and Recap

[Matt Fredrickson]

Hey all,

It’s been a bit over half a year since AstriDevCon last year.
Unfortunately I wasn’t able to attend, but I wanted to review the
notes and see how everybody in the community is doing with any items
they wanted or committed to, as well as report progress that has been
made from Digium’s perspective on areas it has engaged upon.

In conjunction with that, the cutting of the 14 branch is fast
approaching, coming with all restrictions associated with merging code
into release branches.  Consider yourself warned :-)

So, referencing
https://wiki.asterisk.org/wiki/display/AST/AstriDevCon+2015, I’m just
going to list the actionable items of interest that I was able to pull
from the notes from AstriDevCon.  If I missed something it wasn’t
intentional.  I also added notes next to the things that I have seen
completed or in progress in my short time back in Asterisk land.  As
mentioned, feel free to reply with things I missed or statuses that
are incorrect.

>From Loway/Lorenzo Emilitri’s presentation:
app_queue skills based routing patch slide
- patch at https://github.com/pascomnet/asterisk_sbr
- Has this been submitted to be merged?
Music on hold slide
- no patches currently posted so not sure what kind of progress was
made on it.  Comments?
Transfers slide
- Couldn’t find much as far as specifically actionable items
WebRTC integration - Want easier way than browser based SIP clients
- Just using Asterisk/chan_sip not sure if things are better yet
Remote file linking (HTTP):
- Implemented by Digium/Matt Jordan
Remote audio streams
- status unknown
Deploy a production box via ARI
- Can push PJSIP configuration and other things via ARI now -
https://wiki.asterisk.org/wiki/display/AST/ARI+Push+Configuration

>From Ben Klang’s presentation:
MRCP via ARI
- status unknown
DTMF Matcher (with SRGS)
- status unknown
SSML Parser
- status unknown
Media Playback Optimizer
- status unknown
Arbitrary Tone Detectors
- status unknown
Asymmetric Audio Bridges
- status unknown
Remote fetch/store audio
- Matt Jordan/Digium added support for this in master/14

>From Torrey Searle’s presentation:
New Function added to chan_sip: SIP_SDP_OFFER – returns a csv of
codec/bitrate/type
- status unknown
New Channel Variable SIP_CODEC_OUTBOUND_ORDER – allows a csv of codecs
to be specified, controls the order of SIP Codecs on
the Outbound Leg, codecs not allowed by the peer are removed
- status unknown
If a call is bridged copy the codec order from the B leg to the A leg
in the answer
- status unknown
If a call is bridged and there are common codecs between the B leg and
the A leg, remove all not supported by B in A's response
- status unknown

I had a harder time sorting out Sean McCord’s presentation due to a
lack of slides, but here are a few key points that I found:
Improved ARI documentation of media URIs
- status unknown
Improved ARI documentation of when to use body versus path ids
- status unknown
Improved documentation of how to add new ARI functionality
- status unknown
When websocket connections are lost, issue in that channels hang round
and new channel can enter without notifying anybody.  Behavior
improvement desired here.
- status unknown
Arbitrary tone detection (again)
- status unknown
Add docker file to Asterisk source tree
- Leif Madsen submitted one and it has been merged.

>From other discussion section:
Streaming MOH / audio
- status unknown
Update ARI spec to conform to a newer version of swagger
- some discussion on the -dev list, but no code submitted.

Here are a few additional major initiatives that Digium has worked on:
Publishing extension/device presence to external SIP presence servers
such as Kamailio
- Done - https://wiki.asterisk.org/wiki/display/AST/Publishing+Extension+State
ARI early bridging support:
- Mostly completed

Thanks again for all of your help and contributions.  Asterisk would
not be great without all of you who put in the time and effort
necessary to move it forward.

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA