webrtc
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pion
Add interface to disable DTLS SRTP
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
agreed, TLS should be configurable, coz encryption is not necessary under some circumstances, performance critical i.e.
@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 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
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.
