AstriDevCon 2020 Agenda

Date¶

Discussion items¶

Afternoon topics:

• Outbound ARI discussion (websocket initiates from Asterisk rather from external agent)
• Are there ways to improve bug reports on the public issue tracker?
  • PCAPs
  • Core dumps
  • sipp scenario files
  • Maybe adding a bounty flag on the tracker so that people could figure out which ones have bounties attached.
• What happened to the bridge created event in ARI (not showing up until first channel is added)?
  • Handling on -dev list
• Is there additional information that needs to be added to certain debug/error/warning messages to better understand the original source of error?
  • Consistent identifying attributes on log messages (IAX, chan_sip) - owner information, callId/reference.
  • Add log message per channel driver linking channel_id and channel_name and protocol specific callId/callref
  • Would helper functions help enforce this?
• Group photo
• Discussion around challenges supporting queue strategy changes for dynamic environments
• Adapting codec quality to the network (potentially using RTCP feedback or native means)
  • Feedback to sender to alter sending
  • Handled in codec module (callback exists to provide RTCP information so codec module can adjust)
  • End to end sequence number preservation exists now to help determine gaps for codec implementations
• Improved RTCP stats logging
  • Ability to disable RTCP messages in AMI
  • Would be nice to get RTCP stats in stasisend event (due to not having access in ARI after stasis end)
  • Ability to add to CDR log or CEL event log would be neat. Perhaps using custom CDR log.
  • Better documentation and easier to find
  • Do access to things work in hangup handlers?
• Are there challenges that people have with provisioning fleets of Asterisk instances in the cloud? (missing provisioning APIs, logging interfaces, ...)?
  • Nice to have a solid ARI version of app_voicemail that works well across multiple instances (all playing together well with recordings, metadata, etc)
  • Challenges with the way files are stored with app_voicemail across multiple instances of Asterisk and shared file stores.
  • It would be nice to record files and send them to another server directly from Asterisk (maybe using remote FTP/HTTP storage or something of that nature). We can already playback remotely, why not recording as well?
• What would be interesting to see next?
  • Blind transfer across Asterisk instances
  • Media failover (between Asterisk instances when one fails or for draining of calls from one misbehaving instance to another).
  • PauseRecording/UnPauseRecording support in mixmonitor is missing - recommended workaround is to use mixmonitor stop and then mixmonitor append.
  • Better handling of RTP header extensions in Asterisk. (passthrough or processing of radio related header, potentially also for webrtc and other header extensions)
  • Optimistic encryption support when using DTLS with chan_pjsip