I’m using asterisk 1.2.0 with VoIP (SIP) only. I’m facing some latency delay and echoing problem in the voice converstation.
I know there is one parameter in SIP.conf - tos. Is there anything other than this that can help?
Pls help.
Thanks,
Rajesh.
I’m using asterisk 1.2.0 with VoIP (SIP) only. I’m facing some latency delay and echoing problem in the voice converstation.
I know there is one parameter in SIP.conf - tos. Is there anything other than this that can help?
Pls help.
Thanks,
Rajesh.