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...TDM system. There's the encoding latency, easily understood as the time the device takes to convert the analog signal (your voice) in RTP packets, then there's the transmission latency, inherent to any network, and so on. All those latencies add up to each other, making the total latency go skyhigh and making you hear your own voice delayed by some milisseconds - the infamous echo. Asterisk cannot cancel echo when the call is entirely IP, from an IP phone to another, for example. There's simply no need for that. That's because it's the device's job to cancel the echo caused b...

...for quite some time now, and util last saturday without serous problems. Last saturday however I updated * again, and now the cpu load goes to a 100%. It seems that the sip register is the problem. when I comment out all sip registers there is no problem, but enabling just one register the cpu goes skyhigh. I've seen there were some modifications to the chan_sip.c. Can anyone confirm the problem? Regards Andre -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050620/d1f42bef/attachment.htm

If I connect to a provider using iax, and that provider connects to his provider using only sip, the provider I am connecting to isn't going to be able to bridge the call and drop out of the media stream correct? If I'm understanding how bridging works, you lose the ability to have the media stream going directly between the two endpoints of the call with most of the providers out there

...to locate local IP address for RTP stream We were searching for clues on this matter, but there doesn't seem to be a lot of information on H323. Could someone help us in understanding what the problem is, and how we can fix it? Regards NYAMUL HASSAN ======================================= SkyHigh Communications LLC. 2861 Themis Street Cape Girardeau, MO 63701