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?
Atenciosamente,
Vinícius Fontes
Núcleo de Tecnologias Convergentes
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brasil
+55 54 2104-7000
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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.
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Kind Regards,
