Handling rtcp packets

[Sheel Patel]

Hi,

Question regarding how to handle rtcp packets for webrtc?

1. When I capture audio packets, rtcp packets sent by webrtc client are decoded in wireshark. Does this mean the RTCP packets are not being encrypted while multiplexed with RTP?

2. Did anyone made a call between webrtc client and non-webrtc client (x-lite, linphone etc) ? How do you handle rtcp in such case, when rtcp-mux is being enabled?

Pointers to draft/RFC will be helpful.

Thanks

[Christoffer Jansson]

AFAIK RTCP and RTP headers are not encrypted, is that what you are referring to?

RTP Draft: https://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-26

RTP/SAVPF draft: https://tools.ietf.org/html/rfc5124

/Chris

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/Chris

[Sheel Patel]

Hi Chris,

Yes, headers are not encrypted. But I am seeing even the payload for rtcp is also not encrypted what is sent by webrtc client from chrome. (Basically it is not SRTCP but RTCP packets)

My task is to bridge the rtcp packets between webrtc and non-webrtc client. I am observing the decryption action is not required for rtcp packets when they are multiplexed.

Any attribute indication in SDP which triggers whether RTCP should be encrypted or not.

Thanks

[Sheel Patel]

It looks like the Authentication failure while decrypting srtcp packet.

I was being mislead by wireshark decoding the SRTCP packet and showing all the fields.

But, looks like packets are being dropped while decryption process. which gives Authentication Failure and Replay Check Failed - index too old.

Any pointers on how to debug?