Overview of plan to land webrtc code from Alder to Mozilla-Central:
3rd tranche: in alder; each can land separately. Target date: Sept 15.
libjingle (very small subset - mostly sigslot - see if we can move into
ipc/chromium).
These depend on sigslot (libjingle):
SCTP/DataChannel (netwerk)
transport service (ICE/TURN and p2p transport from EKR)
Opus
libsrtp
libyuv
update of webrtc.org code
more tests
4th tranche: Target date: Sept 30th.
signaling (sipcc)
This is large, on the order of 200-250K lines.
peerconnection DOM work
update of webrtc.org code
more tests
---------------
Reviews needed:
We will NOT be line-by-line reviewing imported code. We will be importing
versions chosen for stable snapshots by Chrome in order to best leverage
Google's testing and security work. This also will let us watch any
important bugfixes and security fixes to those 'stable' pulls. Updates on
m-c after landing will be pulled from Chrome stable revs, but with perhaps
a bit more examination of the ongoing changes as the size of the patches
goes down.
Updates to third-party code (libyuv, libsrtp) to be handled
in conjunction with webrtc.org updates
We will be doing normal line-by-line reviewing of code we wrote
and modifications to the imported code.
Security reviews: To be negotiated with Security team; done in phases
and leveraging Google's work.
Most likely soft spots will be DOM and signaling. Maybe DataChannel.
We need to tie into Google's security team
We'll need protocol fuzzing
Avoid too much overlap with Google's testing
cdiehl has experience with fuzzing
Protocols available for fuzzing:
RTP/RTCP
SRTP/SRTCP (libsrtp)
DTLS
SCTP/DTLS
ICE
STUN
TURN
VP8 and OPUS packetization (may not be much there)
NetEQ/etc (fuzz by modifying jitter and loss)
JSEP/SDP (huge possible space)
User privacy protection and controlling the attack surface of the
browser will be important considerations.
Opus support in WebRTC
Reviewers: jesup, derf
DataChannels
Note: most of the base library is already reviewed.
Reviewers: mcmanus/biesi (netwerk/sctp), jst/peterv (DOM)
mtransport:
Reviewers: mcmanus/biesi/jesup, security
Review libsrtp and libyuv.
Note: pure import, no line by line review
Reviewers: ekr, derf, jesup, graphics team member?
Signaling: (JSEP/SDP)
This is sipcc - ~200K lines, and a fair amount of modifications
Also, a fair bit of code was added to sipcc after it was
open-sourced and before it was imported from ikran. That
code should get higher scrutiny.
There is lots of "dead" code on paths only used when SIP is
enabled, which it is not in the initial landing. When/if we
enable SIP, we'd want to give once-over review of those pieces to
make sure we haven't violated any invariants of the code with
our other modifications.
We may want to spread the review load wider here, and we should
try hard to break this into separate reviewable pieces.
Some parts are already being reviewed by ekr as mods are made,
and may just need a roll-up review of the final state.
We also are trying to get an engineer from Cisco's SipCC team to
review the mods to the code. (Also, several of the Cisco engineers on the
WebRTC team have worked with SipCC in the past.)
Reviewers: jesup, ekr, derf, mcmanus(?), biesi(?), security
PeerConnection DOM
Reviewers: jst, peterv, khuey?
Updates: We'll take updates from 'stable' branches of Chromium's webrtc
pulls, by pulling the same changesets. We should watch for changes
applied after they're moved to Chrome, and individually review those.
--
Randell Jesup, Mozilla Corp
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