VoIP issue

Unanswered Question
May 11th, 2009

Hi All,

we have SIP service provider for our CME. some incoming calls to CME are one way audio but some work fine. I did "show voip rtp connection", and in the one way audio call, I saw CME chose the loopback interface as source RTP stream, which has private IP address and it's not routeable over the internet, that's why it's one way audio. In the good calls, CME chose the gi 0/0 interface which has public IP address and there is two way audio. How can I force CME to use only Gi 0/0 as the source RTP stream?

thanks

Alex

I have this problem too.
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Nicholas Matthews Mon, 05/11/2009 - 16:24

Hi Alex,

For SIP:

voice service voip

sip

bind all source-interface g0/0

Please note that ALL SIP traffic must go through G0/0 once you bind it. This means SIP CME and all session targets from other routers.

hth,

nick

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