SIP plugin RTCP receiver report for packet loss

[infini...@gmail.com]

Hi,

In our setup, we have WebRTC clients connect to the Janus SIP plugin and the SIP plugin talks to Asterisk. In this setup, if there is packet loss from the client to the Janus SIP plugin, does Asterisk see the same packet loss? In other words, would the RTCP packet loss receiver report first come from Asterisk to the SIP plugin and then Janus would report the same to the client? The premise of the question is with Opus audio if the client would know about the packet loss to start using FEC.

Thanks,

Terry

[Lorenzo Miniero]

RTCP from clients should get to Asterisk via Janus, but the other way around it won't. We terminate RTCP, so any attempt for a plugin to send a SR/RR to a client would be discarded by the core.

L.

[infini...@gmail.com]

Thanks for the response.

Since the RRs generated from Asterisk don't make it to the client, what would have to Opus audio that relies on the fraction lost and packet lost to do FEC? Does it mean the client may not encode FEC or not much of it because it doesn't know about the packet lost at the destination? What fraction lost value gets reported in the RR from the plugin back to the client? Is there a solution here to bridge the client and Asterisk through Janus for Opus audio so the FEC can get used?

Thanks.

[Lorenzo Miniero]

Janus sends RR about its own leg with the client, so FEC may be enabled only if that's where packets are lost. You may want to deploy Janus instances closer to Asterisk, so that the connection between them has less chances of losing packets than the one between Janus and clients.

L.