We are installed the Genesys Application in India for US Calling. We are using the Audio Code Mediant 1000 for Media Gateway and using the G7111A for codec. In some time, the voice has been broken and also conjunction in the voice.
The T-Server logs shows the connection established to the particular number. where can i check the issue and which logs shows exact error value.
Please help to arrest the issue.
Thanks in Advance
Are you talking about SIP Server from Genesys, right? If so set the logs for TServer on full mode to be able to see SIP Messages. Also check your network latency as this may cause the disconnection of the call.
Yes. we are using the SIP Server. and also we are enabled the logs also. In the logs, there is no error.
May be error related to codec or network? how to identify the codec or network error on the particular issue?
Sundaraas,
Do I understand you correctly that call (SIP session) is active but voice stream (RTP) becomes “corrupted” (parties cannot hear each other)? If that’s the case then you would need to check quality of your network, average/peak load etc. I would say that SIP Server log won’t be really helpful here as SIP Server isn’t involved in handling of RTP stream at all. You would need to check AudioCode gateway’s log and probably try to get some statistics (dropped packets etc.) from your network routers.
R.