Providing just "120" as a jitterbuffer is not a good idea, because it just specifies the initial size of the jitter buffer. We changed it to "100:250" which uses 100ms jitterbuffer as a default, but 250ms as a maximum jitterbuffer size. This is big enough even for bad internet connections. However it is also small enough to always provide minimum delay.
Providing just "120" as a jitterbuffer is not a good idea, because it just specifies the initial size of the jitter buffer. We changed it to "100:250" which uses 100ms jitterbuffer as a default, but 250ms as a maximum jitterbuffer size. This is big enough even for bad internet connections. However it is also small enough to always provide minimum delay.
Use the presence_data field to annotate channels with a filterable
identifier that allows us to differentiate SIP.js channels from SFU
ones.
The motivation: allow metrics exporters/instrumentations/etc to
generate comparison metrics (eg.: mediaStats, usage) between the default
bridge and the experimental one without having to do multiple or overly
verbose json_api or mod_command/fs_cli calls to filter channels out.
This new dialplan rule filters calls originating from bbb-webrtc-sfu via SIP
user agent parsing. The default bbb-webrtc-sfu UA is "bbb-webrtc-sfu".
A new dialplan rule is needed to force RTP auto-adjustment for calls originating
in bbb-webrtc-sfu (rtp_manual_bugs=ACCEPT_ANY_PACKETS).
That is due to the fact that bidirectional mediasoup bridging is done via an
RTP/AVPF endpoint which does not use ICE. FreeSWITCH arbitrarily blocks off auto
adjustment for AVPF profiles (presuming ICE), so it needs to be forced otherwise
the bridge won't work properly in all environments.
Bridging mediasoup and FS via WebRTC (which would circumvent that) is currently
not an option due to the fact that FreeSWITCH doesn't handle STUN role conflicts
properly (and there will always be a conflict since the initiator is controlled
and FS always defaults to controlled)
Briding mediasoup and FS via plain RTP/AVP (which would also circumvent that) is
not an option right now due to the fact that FreeSWITCH doesn't make ssrcs
public in signaling for RTP/AVP profiles. mediasoup needs the remote ssrcs.
This could work by pre-generating a ssrc in bbb-webrtc-sfu, signaling it via a
SIP header and then specifying it in the rtp_use_ssrc channel variable in FS,
which would allow us to shim the ssrc in FS's answer in bbb-webrtc-sfu.
Maybe in the future.
- switch SIP ports in vars.xml:
- Internal to port 5090 from port 5060. 5060 is assigned to external sip profile
as bbb-voice only connects to 5060. Tried fixing bbb-voice but can't figure it out.
- Use 5090 instead of 5080 since 5080 is used by Red5.
