On Sat, 2005-01-29 at 20:20 +0100, Zdik Kudrle wrote: > I called from my home thru Asterisk to my Cellphone. I've picked up the > phone and set up input channel instead of microphone my TV card. I started > the TV a listened to the latency cellphone<->TV. Then I said something to > phone and listened to the latency cellphone<->Speakers. Simple, but it > works for me. Today I did the same thing without cellphone - somebody was > sitting in the office and answered local call (no calling out, just intra > PBX call). He sent me a message using ICQ and that exact moment said > something to the phone and vice versa. The result was pretty same using > the cellphone...
Alright, no nifty network latency measurement then :) It's basically what I do all the time too, either echo tests with * or having two clients call each other. > I don't think my soundcard has latency problems (Live!5.1). The really > strange think is that latency is _increasing_. It means that somewhere all > the data must be stored but I've got no idea where. Maybe I'll run some > crashtest - one part or other will crash with out of memory... :-) I should have mentioned that I'm on Linux. It's not a card issue actually (I have a SB too), but rather how driver buffers are set up by the client application. I found that it makes a huge difference with respect to latency on my system. Regarding the 'increasing' point, that's really something I can't confirm. Usually, my latencies are constant, or huge at the beginning and decreasing afterwards. So I guess something's definitely out of the order in your case. > This sounds promising, I'll give it a try. Possible only if you're on Linux, too, since gnomemeeting doesn't run on Windows. Sorry I didn't mention that, but sometimes I forget that there are other OSes out there. Assumed that you're on Windows, I'd still try other softphones like XLite or SJPhone. Especially XLite seems to be pretty good. It's a SIP phone, and in my experience many latency issues are due to IAX or it's implementation. Just today I experienced them on native bridging, and although IAX is marketed as VoIP swiss knife protocol, especially regarding softphones I get best results with SIP and H323. > Sorry to disappoint you, I'm not that kind of network magician.. Me neither, that's alright. Good luck. Regards, Bruno. _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
