Pjsip + Webrtc First second mute

Hello guys,

I have a enviroment with asterisk version 16 and pjsip(webrtc), I have a leatle problem that when I make a call throw originate and send a call to a queue or Dial directly the first Hello of the client is mute in webrtc after almost 1 second the audio is established normally it just happen and a external pjsip(webrtc) in a internal call the first Hello always is normal.
When use a outgoing call throw directly in webrtc it is going normally but When a made a call from originate and after answer send to pjsip(webrtc) always the first Hello of the client is mute and the second word established normal, I compare in the same enviroment with sip.conf directly on the softphone always the first hello is OK, I don’t know if is possible to do some thing because in a local network always work fine, I tried to do many thinks but without result, some one can help me?

ICE negotiation has to complete in order to find a viable path, and then DTLS negotiation has to complete as well. I’d suggest taking a packet trace to examine what is actually going on to determine if that is the cause. You can also do “rtp set debug on” to see how long it takes before it starts sending media out using ICE.

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