[OpenSIPS-Users] One Way Audio

[Ross Beer]

I have a server located on the internet running opensips and asterisk. When registering directly to asterisk I can perform echo tests and make calls.

 

If I register to Opensips and use the load_balance there is one way audio. I can hear sounds coming from the asterisk server but sound from the soft phone does not reach asterisk. I can confirm this when looking at a rtp debug on asterisk.

 

I can see that traffic is passing from the soft phone when performing a wire shark trace to the server and it also shows that some RTP packet are being passed out and back into my local address. This does not happen if I register directly to asterisk.

 

Any advice you can offer would be appreciated.

 

Opensips shouldn't effect the RTP if it only load balances?

 

Thanks,


Ross
 		 	   		  
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