Hi All, I am seeking input from anyone who may have seen a similar configuration and dealt with similar issues to what I'm experiencing.
Configuration: - 2 sites (site A and B) - Asterisk 1.2.23 on each site (Trixbox) - Internet 512/512 symmetric at each site, dedicated to VOIP calls only. - IAX trunk between the sites, with data travelling across the 512/512 Symmetric link - PSTN inbound/outbound via a Sangoma PCI FXO card. The required configuration is inbound calls at either site need to be answered by a reception at Site A. Calls coming in via PSTN to Site B, will result in a SIP extension at Site A to be dialled and answered. This will result in an active channel between site B's asterisk server, and the user at Site A. If Site A transfers that call *back* to site B, this will result in another call leg being established to the user at site B. Every RTP packet will travel: - in via PSTN @ Site B - across 512/512 DSL link to Site A's asterisk server - back across 512/512 DSL link to user at Site B We are noticing jitter and voice quality problems. A call can degrade in quality over time. We are using G729 for the voice codec. Can anyone suggest further debugging I can do to determine the cause of voice quality degradation? Is there a way I can configure the asterisk servers to not communicate the RTP traffic across the DSL links and back again? Any suggestions will be much appreciated. Regards, Chris Bennett _______________________________________________ --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
