This is not all of it as it is not possible (or at least I do not know
of a way) to get all suggestions for a repo/project. Did this via loop
searching mainly and taking the modernize suggestions.
* Make new path for signalling v1.5 support.
To be able to support newer clients to interact with older servers, move
signalling v1.5 to new path (`/rtc1`). On the new path, `join_request`
is required and single peer connection is used.
With the existing path `/rtc`, single peer connection is still supported
if `join_request` is used.
Newer clients connecting to old server should follow
1. Try new path WebSocket
2. If that fails, try new path validate at `/rtc1/validate`.
3. If the above gets a 404 which will happen with older server, revert
back to old path and signalling 1.0.
Open to suggestions on path name.
* test on both paths
* change path from /rtc1 -> /rtc/v1
* test all rtc service path combinations
* Make data message naming a bit more consistent.
OnDataPacket and OnDataMessage half-way in the chain made it confusing
(for me at least). Use same name throughout.
API still uses SendDataPacket, but that is not harder to read.
* test
* Use published track for model access in data down track.
No need to pass in the model.
* implementation type assertion
* remove redundant log key
* one more type
* Close interface method
* clean up redundant log key
* remove unused method
* some interface changes to support data tracks in replay
* correct mock signature
* Add support for RTP stream restart.
When an unhandled packet is encountered, try a restart sequence.
Restart happens when 5 packets with contiguous sequence numbers and same
or increasing time stamps are received. Note that this does not work for
B-frame type of scenarios, but that is true for receive path handling
even before this. As WebRTC does not use B-frames, it is fine. But,
needs to be looked at again if B-frames are necessary.
It is controlled by a config that is disabled by default.
* clean up
* debug log
It is possible that the stream stops just after start and
restarts much later introducing a large gap in sequence number.
That could look like an unhandled case because the wrap back handler
does not have enough packets yet.
Let other checks based on time stamp gap take effect and only if that
also leaves the sequence number unhandled, drop the packet.
* switch participant callbacks to room to listener interface
* mage generate
* clean up
* clear listener
* clean up
* use interface in up data track manager
* tweaks
* Paul feedback - should reduce the diff as this keeps the room handlers as is except making methods for a couple of anonymous handlers
* clean up
Packets not being forwarded were getting included in forwarding stats
calculation and skewing the measurement towards a smaller number.
The latency measurement does not include the batch IO of packets on
send. With a 2ms batching, that will add an average latency of 1ms.
* if RingingTimeout is provided, deadline should be set to that timeout.
This is because the SIP bridge will not return until RingingTimeout
which may be longer than the 30 second default deadline.
* handle Deadline being "before" timeout.
Currently, the signal requests are counted on media side and signal
responses are counted on controller side. This does not provide the
granularity to check how many response messages each media node is
sending.
Seeing some cases where track subscriptions are slow under load. This
would be good to see if the media node is doing a lot of signal response
messages.
publisher peer connection.
While cleaning up during single peer connection changes, unintentionally
removed handler.
Also, another small change to log first packet time adjustment after
increment.
