I am facing an issue where the voice of the called person is not reaching the callee. This has happened for a few calls.What are the possible reasons for this?
I have checked the codecs being received, and it is ok and no problem with that. What are the other factors I can investigate regarding the matter.
Most common cause is misconfigured NAT, followed by misconfigured firewalls.
But if it was for firewall or NAT issues, won’t it reflect on all of the calls?
Not if you have spot rules for certain port numbers.
Generally, you need to provide logs showing the SDP handshake and whether or not Asterisk is receiving the media streams.
Asterisk is receiving the media streams as I am sniffing the packets received and there I can see that the media streams are being received, but the caller is unable to hear listen to them.
Is Asterisk seeing them, according to rtp debug, and does it thing it is sending them? Does the sniffing indicate it is actually sending them? Where are you doing the sniffing?
Yes, I can see rtp streams being sent over from asterisk to the service provider.We have a proxy server, between asterisk and service provider, and there we are sniffing the packets being received and sent.