Media/WebRTC/2012-10-09
From MozillaWiki
10/9/2012, 9am Pacific:
Notes:
1) Our next big push will be
- Deal with critical bugs: crashes & security problems, then major functional defects
- Get all the code to the point where we can pref gUM and WebRTC on.
- Goals for Firefox 19: (1) gUM preffed on by default (2) Interop with Chrome M24 (3) Pref on WebRTC (stretch goal)
- In prep for those:
- How do we want to use bugzilla? (We will continue to use blocking vs nonblocking)
- How do we want to use Alder? Where should we do our work going forward?
- three possibilities:
- everything is a bug, straight to inbound
- use alder as staging, but strict r= rules like inbound; merge to m-c directly periodically
- use alder as a sandbox, keep as patches to land on m-i
- Seems to be consensus towards option 1 (normal mox dev process), with collaboration via git or shared patch queues.
- three possibilities:
- Make sure all issues are in bugzilla (reminder)
2) Where are we on critical bugs for Firefox 18?
- Take a look at bug list (at the bottom) now or at the end?
- review in triage tomorrow
3) gUM
- Where are we on gUM functionality both in terms of code to support the UI and blocker bugs for preffing gUM on?
- There are 22 blocker bugs (based on my search in bugzilla this morning).
- Anant refactoring media code, should land this week
- Most important: Merge two media streams into one (Bug 758391), If you stop the audio stream, you have to restart the browser to get back the audio stream (Bug 773646).
- (jsmith) gUM - regression on trunk when viewing a video stream off of anant's page - I'm seeing crash volume on https://bugzilla.mozilla.org/show_bug.cgi?id=799191
- For Bug 799191: Fix in the bug up for review!
- What happened to dom/media/test?
- Anant's test pages should detect if older browsers are used to access them
- set media.navigator.permission.disabled=true if you want it to work today.
- Anant refactoring media code, should land this week
- There are 22 blocker bugs (based on my search in bugzilla this morning).
- UX design (Latest design is in Bug 729522)
- Missing persistent notifications
- Missing icons
- Buggy for non-sunny day, multiple device situations
- BIGGER: How much do we have to redo to address privacy team's concerns? Should we try to have a meeting with Boriss and Monica this week?
- We need to know what they're looking for
- Do we need persistent?
- Have a meeting
- We need to know what they're looking for
4) Interop with Chrome & What do we need from Google (this week)?
- What do we want to push during our hangout with Google tomorrow?
- interop
- not good for spec if we don't interop, will get press
- interop
- Soon we want to identify what parts of the spec we need to push forward (revisit the existing spreadsheet this week)
- Opus story
- technical issues aren't big
- really licensing
- ICE
- need to try interop
- DTLS
- NSS access in sandbox (they believe they have a solution)
- expect patchset by end-of-week, land early next week
- Overall: close to being able to try interop
- ekr has a testbed
- talk about a command-line testbed
- Opus story
5) Testing
- Bug Triage Meeting: Wednesday at 2:30pm (a different time this week, back to its regular time next week)
- From jsmith (he can't make today's meeting) -- In the full stack of webrtc - what's ready to test vs. not? I saw a bunch of stuff land, but just want to clarify what I can start hammering.
- (jsmith) Priorities and tests defined in https://etherpad.mozilla.org/automation-webrtc. Feedback welcome!
- Blockers or Issues to Automation based on "ideal" smoke tests needed:
- Audio & Video Joint MediaStreams not implemented
- Stats not implemented
- Not sure if 'start with video connection, add audio later' is supported right now or not
- Blockers or Issues to Automation based on "ideal" smoke tests needed:
6) Discuss any current blockers for people or new items that haven't been discussed
- None mentioned
BUG HIGHLIGHTS:
CRASHES
- Thread leakage [Bug 798323]
- WebRTC Assertion failure: mIceState != kIceGathering, at media/webrtc/signaling/src/peerconnection/PeerConnectionImpl. [bug 791423]
- PeerConnectionImpl::AddStream [Bug 791270]
- NrIceCtx::GetGlobalAttributes [Bus 791330]
- SetLocalDescription [Bug 791278]
- WebRTC crash [@sipcc::PeerConnectionImpl::Initialize] [Bug 791165]
- Input validation on AddStream [Bug 791270]
- WebRTC Assertion failure: mod != NULL, at pk11slot.c:1766 [Bug799419]
FUNCTIONAL DEFECTS
- Support arbitrary length SDP [Bug 798873]
- Disable incoming audio when transport fails [Bug 798680]
- Locks for peerconnection [Bug 792175]
- Audio latency [no bug]
- Enforce state transition rules in SIPCC [Bug 784519]
- Can't get audio twice in a row [Bug 773646]<---
- Can't merge streams [Bug 758391] <------
- Whatever defects are uncovered by Chrome interop
MISSING FEATURES
- Implement rtcp-mux. Currently, we have support for this in the SDP but it's not implemented in the transport layer [Bug 777524]
- Implement bundle [ehugg, emannion, [Bug 784491]
- Allow audio w/o video, unidirection audio/video [Bug 784515]
- Allow >1 of each type of media stream [Bug 784517]
- Send trickle ICE (lower priority) [Bug 784161]
- Implement/wire up TURN. This is already in nICEr but may be obsolete/broken [Bug 786235]
- Implement per-flow STUN configuration [Bug 786236] [UPLIFT]
- Audio/video sync [???]
- Identity support [???]
- Constraints [???]