There are two approaches you can take: 1. Use the protocol level tools that support this sort of stuff 2. Monitor an audio path
1. Protocol Support ------------------- If whatever is on the remote end supports RTCP (real-time control protocol), you can use monitoring tools specifically for that which will set up a call and measure all sorts of things. For example, the open source pjsip stack comes with a sample program for doing just that: http://www.pjsip.org/pjsip/docs/html/page_pjmedia_samples_siprtp_c.htm Here's a snapshot of the output between my desktop computer, and a call I've placed out an FXS port on a Sangoma card running with asterisk: (ip address is my computer's adderss, sip:192.168.1.10 is the remote asterisk system) $ ./siprtp --ip-addr=192.168.1.100 sip:192.168.1.10 10:08:49.570 os_core_unix.c pjlib 0.5.10.3 for POSIX initialized 10:08:49.571 siprtp.c SIP UDP listening on 192.168.1.100:5060 10:08:49.591 siprtp.c Making 1 calls to sip:192.168.1.10.. Enter menu character: s Summary l List all calls h Hangup a call H Hangup all calls q Quit >>> 10:08:51.709 siprtp.c Call #0 connected in 2118 ms l List all calls: Call #0: CONFIRMED [duration: 00:03:48.746] To: sip:phone-server;tag=as731ae06a Signaling quality: got 1st response in 60 ms, connected after: 4558 ms Stream #0: audio [EMAIL PROTECTED], 20ms/frame, 8.00KB/s (9.06KB/s +IP hdr) RX stat last update: 00h:00m:02.560s ago total 11.04K packets 1.82MB received (2.19MB +IP hdr) pkt loss=0 (0.0%), dup=0 (0.0%), reorder=0 (0.0%) (msec) min avg max last loss period: 0.000 0.000 0.000 0.000 jitter : 0.000 0.249 11.375 0.000 TX stat last update: never total 11.04K packets 1.82MB sent (2.19MB +IP hdr) pkt loss=0 (0.0%), dup=0 (0.0%), reorder=0 (0.0%) (msec) min avg max last loss period: 0.000 0.000 0.000 0.000 jitter : 0.000 0.000 0.000 0.000 RTT delay : 0.000 0.000 0.000 0.000 2. Audio path monitoring ------------------------ Check with Simon Rowland. I know how to do it, but it wasn't my idea. On Fri, 11 May 2007, Jim Van Meggelen wrote: > Folks, > > Does anyone know of a decent suite of VoIP testing apps that can be used to > generate reports? > > What I want is to simulate a phone call at 64K between two links for a 24 or > 48 hour period, and then be able to plot the data on a graph. > > I am getting complaints of intermittent static on calls, and while we have a > DS1 to each site, I don't expect it is dedicated bandwidth the whole way. I > want to be able to report on a whole day's worth of traffic and be able to > say something like "at 10:33, for 15 seconds, we had 50% packet loss", or > some such. > > Any ideas? > > Jim > > > -- > Jim Van Meggelen > Core Telecom Innovations > [EMAIL PROTECTED] > www.coretel.ca > 416-425-6111 x6001 > 877-CORETEL x6001 (Canada) > IAX2:[EMAIL PROTECTED]/6001 > www.oreillynet.com/pub/au/2177 > > No virus found in this outgoing message. > Checked by AVG Free Edition. > Version: 7.5.467 / Virus Database: 269.6.8/797 - Release Date: 10/05/2007 > 5:10 PM > > > > --------------------------------------------------------------------- > To unsubscribe, e-mail: [EMAIL PROTECTED] > For additional commands, e-mail: [EMAIL PROTECTED] > >
