Add interface to disable DTLS SRTP

pion / webrtc

Pure Go implementation of the WebRTC API

https://pion.ly

MIT License

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Add interface to disable DTLS SRTP #2273

Open CaptainNeil opened 1 year ago

CaptainNeil commented 1 year ago

Summary

As the title describes, I would like to have an interface to disable DTLS SRTP, just like the enable_dtls_srtp defined in C++

Motivation

To analyze RTP payload.

Describe alternatives you've considered

I'm working on rtmp 2 webrtc, the workflow is as follow：

rtmp receive origin aac data
decode aac to pcm
slice pcm to 10ms pcm list (the opus encoder requires)
encode pcm to opus
pack opus as an rtp package
send rtp package to TrackLocalStaticRTP

Unfortunately, the received sound on the other end is noisy. I saved the opus data before sending it to the RTPSender, and it sounded fine.

So I wanted to use Wireshark to see what happened, but the rtp payload was encrypted.

By searching at code，I could'n find a way to turn it off.

@Sean-Der

HustCoderHu commented 1 year ago

agreed, TLS should be configurable, coz encryption is not necessary under some circumstances, performance critical i.e.

CaptainNeil commented 1 year ago

The problem is similar as #2090

zjzhang-cn commented 1 year ago

agreed

yyyar commented 1 year ago

@CaptainNeil did you managed to solve the issue? I have similar problem doing the same thing (aac->opus->webrtc) I've got choppy audio as described in #2090. I noticed 0 lost packets in webrtc-internals in chrome, but lots of concealedSamples and insertedSamplesForDeceleration.

CaptainNeil commented 1 year ago

@CaptainNeil did you managed to solve the issue? I have similar problem doing the same thing (aac->opus->webrtc) I've got choppy audio as described in #2090. I noticed 0 lost packets in webrtc-internals in chrome, but lots of concealedSamples and insertedSamplesForDeceleration. @yyyar I have solved this problem. My problem is the noise caused by the rtp timestamp. When I pack OPUS into an RTP packet, I did not increase the timestamp according to the audio size. Therefore, noise will be generated when webRTC rendering

GodKingGitHub commented 7 months ago

absolutely agreed

Sean-Der commented 1 month ago

I am in support of this!

What someone should do is add support for the NULL cipher to pion/srtp. Then via the SettingEngine both sides can request it.

As long as this requires the developer explicitly requesting it (via the SettingEngine) it would be great to have.

sirzooro commented 2 days ago

I am working on some changes to srtp module now. I will add NULL cipher there too.