[asterisk-users] RTP packets dropped

Matt Riddell matt at venturevoip.com
Tue Jul 15 20:28:48 CDT 2008


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Vinícius Fontes wrote:
> As RTP packets have a sequential number, is there some logging/debugging option in Asterisk to monitor how many packets have been lost on a SIP call?

You could use rtcp stats if the endpoints support it.

- --
Kind Regards,

Matt Riddell
Director
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