I find smokeping (http://oss.oetiker.ch/smokeping/) to be very handy as a longer-term monitor of ping times. I have a section devoted to SIP peers on my home machine (see http://home.bod.org/smokeping/, click on the 'SIP Peers' graph). Darren raises a good point, though - those numbers are for the SIP hosts, not the media gateways.
The numbers here are for a linux box on a fast, low-latency residential wireless connection, located in San Jose, CA. You may recognize a couple of the hosts as UK service providers (I'm an English ex-pat). -- Paul Darren Sessions wrote: > Another thing you may want to do is try a simple ping test to the far > end host. While this may not always be a reliable way to test lag > given that the far end maybe just a proxy and your RTP may be > terminating to another device, it still should give you a good idea > what your lag times are at least on the signaling end of things. You > could also do a traceroute to see how many hops your having to jump > through as well. > > You could use a tool like ngrep to actually see the sip signaling and > copy out the media gateway from the SDP if you really wanted to, and > do a ping on that. > > I've done extensive work with international voip origination and > termination, and typically I haven't had any problems unless it's > going over satellite (lag) or there is a problem at the far end > (usually pdd or quality issues). > > If things keep up, I'd even consider running top during a call to see > what kind of utilization your local server is at just to make sure > something isn't wrong there either. > > Hope this helps, > > - Darren > > _____________________________ > > [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]> > http://www.darrensessions.com > _____________________________ > > > On Aug 18, 2008, at 10:41 AM, Nikhil Nair wrote: > >> Hi, >> >> I'm running a small Asterisk server in the UK, just for personal use. >> I've been experimenting with various VoIP providers for international >> calls to PSTN numbers, particularly to the US (often California). My >> results, to date, have been very variable indeed, so much so that I'm >> considering getting a suitable card and using the PSTN. >> >> I have found a VoIP provider with an excellent reputation, and it gives >> very good quality. However, I seem to get quite a bit of delay at >> times, >> enough to make conversation awkward. As the setup at the far end was >> not >> completely trivial, I'm not 100% sure the problem was in my connection, >> but I'd like to test that. >> >> Are there any US numbers I can call to get an Asterisk-style echo test? >> Ideally, a California-based numnber, so I can try to call it from an >> ordinary PSTN phone here, and compare calling via VoIP, and see if >> there's >> an appreciable difference in the delay/quality. I don't anticipate >> using >> this for very long, so it doesn't necessarily need to be a free service. >> >> Failing that, does anyone have access to a US-based Asterisk server >> which >> would allow me to make connections to its echo test? Presumably, if >> I had >> this, I could rent a PSTN number from a US-based provider, and point >> it to >> the appropriate SIP/IAX address. I expect my total usage would be >> just a >> few minutes, though having the facility available for a few weeks >> would be >> helpful, to allow me to play around with various options. Again, I'd be >> willing to pay a modest amount for this. >> >> Thanks in advance for any suggestions! >> >> Best wishes, >> >> Nikhil. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
