That pjsip suite looks pretty good. I'll play around with that for a bit. Thanks much.
Jim > -----Original Message----- > From: Simon P. Ditner [mailto:[EMAIL PROTECTED] > Sent: May 11, 2007 10:22 AM > To: Jim Van Meggelen > Cc: [email protected] > Subject: Re: [on-asterisk] Network testing for VoIP > > 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] > > > > > > > No virus found in this incoming message. > Checked by AVG Free Edition. > Version: 7.5.467 / Virus Database: 269.6.8/797 - Release > Date: 10/05/2007 5:10 PM > > 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
