I have issue with call silence or air dead like during call conversation the remote party suddenly become silent while the channel is still up. I had captured pcap and found that in such calls there is no RTP packets After the call is established but strange thing is I still have call recording on asterisk via MixMonitor app.
The SIP call flow is below for such calls. I am guessing this may be due to directmedia between the endpoint and GW but how then Asterisk captured the call recording via MixMonitor.
If MixMonitor is still recording then media has to be coming from somewhere… what is the Asterisk console output with “rtp set debug on” on as well? What is the network topology and layout, where are devices and what’s on each end?
Thank you for the response. I have below setup.
End Points: mixed of Zoiper/Bria/MicroSIP
Asterisk ver: 11.21
DIGIUM G200 gateway for E1.
The network is simple. I captured the traffic between asterisk and gateway. I am having issue that remote party call goes in slience while channel is up. I believe this is due to rtp packets losing or incorrect sequence/rtp timestamps. Any recommendation or suggestion will be appreciated.
As I understood, you have the following topology:
Phones <—(SIP)—> Asterisk <— (SIP) —> Digium G200 <— (E1/PRI) —> PSTN.
There might be many causes of this issue and many things are still unclear:
- Once the voice suddenly disappears, does it restore after a while or you need to redial?
- The issue may appear at the remote end. Have you tried different users / directions? Does the issue persist for all of them?
- If #2 is “yes”: set a local loop on E1 interface of your gateway and reroute local calls via E1. Does the issue persist? If not, it is not he local network or equipment that causes it.
- If #3 is “yes”: set directmedia=no and run tcpdump -w (file).pcap on Asterisk server. Do you see any RTP traffic towards called party once the silence happens?