-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Yitzhak Bar Geva wrote: > How can one measure the effect of NAT traversal packet loss? > We currently have no solution for NAT traversal for our SIP clients. There > is no doubt that packets are getting lost. What is not clear is how much > damage this does. On the face of it, everything seems fine. Could this be > so? Perhaps we're suffering a degradation in quality or our call setup times > could be improved. How can we measure this? > What's the simplest method of preventing packet loss due to NAT traversal in > a SIP environment?
NAT is unlikely to cause a percentage of packets to get lost. Normally you'd have one way audio if NAT was causing a problem (i.e. 100% packet loss). The only other situation in which it might happen is where the NAT router decides to close a port mapping (thereby blocking incoming calls to the customer's device). But if you're looking for packet loss there are a number of other things to check first. I wouldn't do VoIP across the WAN without at least some packet shaping but hey. - -- 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://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss) -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHHUC+DQNt8rg0Kp4RAv0uAJ9Q41eQ+7RuqzFvgtxEhQOIU0QFggCaAlkD GMVdY/n58wHsciuHihZCCHY= =6L87 -----END PGP SIGNATURE----- _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
