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, Vin?cius Fontes N?cleo de Tecnologias Convergentes Canall Tecnologia em Comunica??es Passo Fundo - RS - Brasil +55 54 2104-7000 Convergent Technologies Core Canall Tecnologia em Comunica??es Passo Fundo - RS - Brazil +55 54 2104-7000
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Vin?cius 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 _______________________________________________ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFIfU7PDQNt8rg0Kp4RAo1dAKCNUKO3NvVKnce7FNk2rI/4D1YfQwCfSXMl T+0EYmctykhpP3he1FCQiPY=EZSI -----END PGP SIGNATURE-----