Error: ssrc already exists in RTP listener

mediasoup libraries
1

Hi,

I am running a PTT scenario.
The app works by opening send/receive transports initially.
Whenever you want to talk a new producer is created along with matching consumers on the receiving end.
When you finish talking I close the producer and the consumers are automatically closed.

Once in awhile, I get the following error: “ssrc already exists in RTP listener” which seems to come from inside the mediasoup workier.

I am not sure exactly how ssrc are allocated so I have no idea how to approach this issue.

Any advice will be greatly appreciated.

Thanks!

2

Could you enable mediasoup debug logs and share the output?

Could also check to make sure you aren’t calling produce more than once on the same producer.

3

Looks like you are creating a producer (this is, calling to transport.produce()) with the same parameters multiple times.

Why are you deleting the producer when the person finishes talking, and then creating a new one again when starts talking, instead of reusing the producer and simply pause and resume it?

4

I think you are both right regarding creating the producer with the same parameter more than once.
I still haven’t figured it out 100% but from the looks of it this could be a race condition in my code that is causing this. Thanks for the advice!

@jmillan, there are 2 reasons for not pausing the producer:

  1. There is sometime a need to switch the consumers, I could not find a way to stop the current consumers and create new ones without generating a new producer entirely.
  2. I noticed that although there is silence there are still packets going through the stream. I want to conserve the battery when the user stops talking for a long time.

If you have an idea how to tackle these 2 issues that would be great because pausing and resuming takes much less time and that would be a far better solution.

Thanks,
Yonatan

5

No idea with “switch to consumers” means. Regarding “a way to stop the current consumers and create new ones”: what is the problem? You don’t need to close and re-recreate a Producer for that. Just close a Consumer and create a new one. I’m afraid there is absolutely no reason for doing that.

In clisen side call audioProducer.replaceTrack({ track: null }). It will make the browser not send any RTP packet, not even silence RTP packets (but just STUN packets from time to time). Then, when talking again, call audioProducer.replaceTrack({ track: originalTrack }) again.

6

Ok, I see your point.
I will try and see if signalling the remote side to close the consumer works (in terms of performance).

Regarding the 2nd point, I am working on native (iOS/Android) currently I am disabling the native track but I will try and see if there is a native API for replacing the track with some null value.

Thanks