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Person Organization Country
Matt Jordan Digium US
Scott Griepentrog Digium US
Gary Hoskins Vectra Software US
Sam Galarneau Digium US
Rusty Newton Digium US
Joshua Colp Digium US
Malcolm Davenport Digium US
Mark Michelson Digium US
Brad Watkins ThinkingPhones US
Alan Graham ThinkingPhones US
Paul Belanger PolyBeacon Canada
Leif Madsen ThinkingPhones Canada
Sean Bright CallShaper US
Andrew Nagy Schmooze US
Nir Siminovich Greenfield Israel
Eric Klein Greenfield Israel
Chad Singer USAN US
Dan Collins USAN US
Torey Searle VoxBone Belgium
Vincent Morsiani VoxBone Belgium
Bryan Walters SchmoozeCom US
Kevin Miller Vantage US
Jason Parker Schmooze US
Rob Thomas Schmooze Australia
Marco Signorini Loway Switzerland
Lenz Em Loway Switzerland
Junichi H Digium US
Jakub Klausa SS7 Tech Poland
Daniel Mierla Asipto Germany
George Joseph Farivew 5 US
Russ Meyerriecks Digium US
Jared Smith Bluehost US
Andy Smith Truphone UK
James Body Truphone Great Britain
Antony Russell Clarotech South Africa
David Duffett Digium Great UK
Matt Frederickson Digium AL
Ben Klang Mojo Lingo US
Darren Sessions Avoxi US
Corey Farrell Raynet Tech US



All times are in PDT

Project Policies (10-11:15am)

  • Allowed changes (LTS/Standard)
    • Standard is okay with test requirements
    • Jared: policy worked really well and everyone seems to think so
    • Does anyone run standard in production?
      • Jared/Daniel: yes
      • Aggressiveness in Standard release is warranted
        • Should we be more aggressive? Particularly in the initial stages?
        • API versioning: when should it apply?
      • Daniel: don't change some things in LTS
        • Data models should not be changed
        • Configuration schemas should not change
        • No breaking changes in LTS release
      • LTS can allow new features, but must be low impact. No breaking features, does not impact the user in a negative fashion
        • Require community notice and sufficient time (1 week prior to cutting RC) to accommodate. If new modules, merge disabled, then enable if approved.
      • Standard: allow more aggressiveness
      • Action: Matt to draft wiki page
  • External libraries (help packagers)
    • pjproject: such a core part of Asterisk. Was embedded. General speaking, do we feel it should be external?
      • Stability is iffy. pjproject is not designed to be an external library. People have to be careful when updating.
    • Jared: generally, yes. RH side is appreciated. Harder sometimes on the end user, probably things we could to to help them.
      • May need better documentation, particularly for end users
    • Asterisk:
      • editline, iLBC
  • Infrastructure
    • GIT!
      • George: really, just do it.
      • Darren: Stash integration with JIRA is nice.
        • but it doesn't bundle notifications
      • Will we do pull requests?
        • Only if it's the only mechanism. Multiple mechanisms are hard.
      • Can we make it easier for contributors?
      • Do we do the Linus model for merging?
        • Paul: is it a developer's job to merge?
        • Sean: not a problem for merging between branches
        • Matt F: It isn't free. There is work involved.
        • Downside of open stack approach: lots of moving parts
          • Leif: you can pick a subset of those parts
      • Action: finish evaluation
  • Testsuite
    • Dependencies are the hardest part. Debian packagers:
      • attest doesn't work (but do we care?)
    • Paul: can we name the test suite?
    • George: trying to get all of the tests to run usually blows up. The documentation is lacking.
    • Torey: versioning of the test suite?
      • Paul: I have a solution to that. It's radical. Remove the tests from the test suite (and put them somewhere else).
      • Version the 'stock' test suite libraries
      • Action: determine what it would take to do that as we move to git
      • Have test suite in docker. Community: provider docker config

Application Server (11:20am-1pm)

  • Deprecate AMI/AGI (Ben Klang)
    • George: configuration
      • "New" configuration - sorcery/config framework
        • Push model of configuration - ARI
      • "Old" configuration - (voicemail.conf, sip.conf, users.conf)
        • Update static files, then do a reload
        • Preferably preserve old mechanism in some fashion
    • Motivation on deprecating it
      • There is no one API. This makes it difficult for versioning. A change to any one makes versioning challenging.
      • If we don't remove something, ARI may make things more painful.
      • AMI/AGI not sufficient by itself; if ARI is sufficient, that'd be nice
      • Can we at least get to the point where ARI is the future?
      • Paul: if multi-interfaces are the problem, why can't AMI/ARI be condensed?
        • Example: AMI/ARI have their own event systems
        • Let AMI use the ARI web socket for its events
          • Would have to solve semantic issues between AMI/ARI event streams
      • Do we want ARI to replace AMI (take its role)?
        • Should ARI become a call control protocol?
        • Lenz: till lots of old stuff in the field. Doing away with all of it will not be easy. A change to ARI will not happen tomorrow unless you control the deployment.
        • Ben: can't just pull it. Just don't want to keep enhancing it.
      • Example: Security events
        • Should they be in ARI?
        • George: we probably do need to have the distinction.
        • Nir: vast majority of community doesn't understand ARI. It needs to be easier in order to replace AMI.
        • Jared: adoption may be slow since it doesn't do everything AMI does.
        • Action: -dev/-users list: what are using AMI for that we can't use ARI for?
        • Leif: we're in a transition, moving from dialplan model to external control model. Probably need external application to be built for us to move completely away from AMI/AGI.
        • Paul: take away apps, and whatever is in the core is what we should care about
        • Lenz: some things are harder in ARI
          • TTS is actually harder (I can use an AGI script for that)
          • Playlists
        • Should we use the dialplan at all?
          • The idea is: ARI should not use the dialplan applications at all
          • pbx_stasis: catch-all for channels
  • ARI Enhancements
    • Play media from remote sources
    • TTS/ASR
      • Leif: write down how you think one would to it
      • Action: Matt to write things down
      • Grammar processor
        • DTMF & ASR
        • Adds latency to application layer today
  • What things does ARI not do (that you would like to to do)?

    • Transfers
      • Need a way to tell the endpoint to redirect to another system
        • This does not exist today
    • Playlists
    • Subscriptions
      • Wildcard: subscribe me to all bridges (or everything)
    • Security Events
    • Have Asterisk initiate the web socket connection (client)

Scalability (1pm-2:15pm)

  • Message bus support (Reddis/RabbitMQ)
    • Externalizing Stasis Message Bus
      • Really want just the state of the system
        • AMI syntax: still clunky, telnet breaks
      • Darren: more interested in scaling
      • Nir: How do I maintain the state of the user, and their location?
        • AstDB (sorcery)
        • Location of a device (and which system it is on)
          • Maintaining synchronization of location across all systems
        • Documentation
          • Jared/Leif: 'sample configuration' that solves X
            • Example: single-system PBX
      • Problem: every scalable system looks different
        • One way of approaching this would be to have a cookbook - defined solutions for defined problems
        • Higher-level view of Asterisk is an ecosystem (with proxies, etc.)
        • Daniel: scaling out is a matter of requirements. Keep the core infrastructure simple, put the load on the client side
  • Simplifying sitting behind a proxy
    • Different identification schemes
    • Registration state dictates endpoint/device state
      • If I have a call from something, assume it exists
      • Have the presence of a call signal that an endpoint/device state exists. Derive from the signalling.
      • Action: Daniel to provide an e-mail describing the problem
    • Need to know system load
      • NEED: CPU usage (Sean Bright and Malcolm: SNMP?)
      • Channel count
      • Memory usage
      • Torey: we send an INVITE once a minute and see what happens
      • Call quality statistics (secondary indicator)
      • Distribution of response times on requests/responses
      • Call quality (see above, again)
      • Darren: two scales - one for concurrent, one for calls per second
        • Caching mechanisms in Linux can be self-defeating. Caching at that call volume can be rough
          • Look at IOStat
  • Simplifying routing/dialing of SIP URIs
  • Federation

Media (2:15pm - 3pm)

  • Codec Visibility
    • Codec control from dialplan
      • PJSIP_MEDIA_OFFER may provide this, would be nice for chan_sip
    • Better control over SDP Offer/Answer
      • Force re-INVITE for codec changes
      • or other things (such as a new media address)
  • BUNDLE support
  • Trickle ICE
    • Need better documentation for WebRTC to aid in testing/dev
    • During PJSIP development, the rapid pace made it hard to follow
    • RTP: have someone act as a sponsor, have someone do the PR
  • RTCP enhancement (Olle)
  • res_rtp_asterisk "crufty"
  • Live call failover???
    • Relatively hard to do
    • May not be the highest priority
    • Other media externalization
      • Paul thinks it would be interesting
      • Daniel: Media processing as a service by Asterisk
        • Nothing advanced, just basic stuff like an announcement or something

PJSIP Enhancements (3pm - 3:30pm)

  • Dial an AoR
    • endpoint -> Aor -> contact1 & contact2, etc.
    • Note We need more documentation for this
    • Disconnect between device state and contacts. Need to know which contacts are in use.
      • There's no guarantee that information in SIP messages maps back to a contact
    • Action: document & explain the problem
  • SIP Outbound
    • Client or Server?
    • May be solved better by DNS support
    • Not much pressure from market for it
    • Table SIP Outbound until DNS is really good
  • DNS Resolver Woes
    • Are we only concerned with Asterisk's PJSIP DNS resolution? Or do we care about the rest of Asterisk?
      • E.164 lookups are really needed (have to shell out to do it today - Rob)
      • XMPP
    • Complaints about current system:
    • Respect TTLs
    • Asynchronous
    • NAPTR
    • SRV failover
    • IPv6
    • Josh: a lot of this is in how you use it
  • Simplifying routing/dialing of SIP URIs
    • sip: - go out specific peer
    • Action: Ben Klang - propose out on -dev list
  • Fix unloading/reloading of res_pjsip

WebRTC (3:30pm - 4pm)

  • How is our happiness level today?
    • Andrew: no way to determine if can_sip or chan_pjsip should be used for web sockets (open bug)
    • Ben: better logging. When it doesn't work, it's really hard to figure out.
    • Better documentation
    • Opus ...
    • ORTC

Data Capture (Phone Home) (4pm - 4:15pm)

  • Upload backtraces
  • Figure out what crashes are most important
    • Switch DONT_OPTIMIZE off by default
      • Darren: Had a large impact in an Asterisk SCF test
      • May want to turn off for Standard Releases, leave on for LTS
    • Opt-out for data?
    • Opt-in for backtraces?
    • Needs a notice that it's on
    • Anonymize backtraces & open source the anonymize
    • Philippe: this is two projects, stats and debugging), keep them separate
    • Define where the data goes to


GIT needs to happen. Need to improve our infrastructure and make it clearer how someone contributes to Asterisk.

Document features better, including scenarios. Make it easier to configure/deploy Asterisk.

Pushing Asterisk into acting more as a media applications server is very important. It feeds into scalability, into making it do more than what it can do today, and affects more pointedly the features in ARI.