I believe there are options for rtp / audio.. Look at pcap play and rtp echo...
Transcoding would be another beast - if you are allowing it Sent from my iPhone 5 On May 22, 2013, at 10:02 AM, Tommy Cooper <tomcoope...@yahoo.com> wrote: > From the little experience I have I do not think that that is a good way of > testing the quality of voice. SIP only initiates and eventually terminates > the call, once that the call is connected, SIP and therefore Asterisk are no > longer involved. Once the call is connected it is assigned to a trapsport > layer protocol such as RTP. RTP is the actual protocol that delivers the > voice call between endpoints. I believe that the setup of your network, QoS, > codecs etc... determine the voice quality of your system. > > > ----- Forwarded Message ----- > From: Mitul Limbani <mi...@enterux.in> > To: Tommy Cooper <tomcoope...@yahoo.com>; Asterisk Users Mailing List - > Non-Commercial Discussion <asterisk-users@lists.digium.com> > Sent: Wednesday, May 22, 2013 3:23 PM > Subject: Re: [asterisk-users] Stress testing Asterisk > > I have a question here. > > How can we test the quality of voice upon increasing the call load? > > Can we try passing a voice file using sipp and record the same in dial plan > record application ? Is this reliable enough to simulate near real world > scenario? > > Mitul > > On Wednesday, May 22, 2013, Tommy Cooper wrote: > Thank you for your help I finally solved this issue. Is it possible that my > setup can achieve 212 concurrent calls, I am running Asterisk on just 1 core > using 3.5 GHz, and 1Gb of RAM? > > ----- Forwarded Message ----- > From: Marie Fischer <ma...@vtl.ee> > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Sent: Wednesday, May 22, 2013 1:16 PM > Subject: Re: [asterisk-users] Stress testing Asterisk > > > On 21.05.2013, at 0:05, Tommy Cooper <tomcoope...@yahoo.com> wrote: > > > Hi, > > I just installed Sipp 3.3 on CentOS 6.3 and all of the calls Sipp is > > generating are failing. I am trying to run Sipp on the same machine as > > Asterisk PBX using the ./sipp -sn uac 192.168.1.115 command. > > Do you have a peer and extension configured for SIPP in your Asterisk > configuration? You also needat least the -s <extension_to_dial> option on > your sipp command line. > http://hasnainali.wordpress.com/2009/03/12/using-sipp-for-stress-testing-asterisk/has > some simple instructions which should get you started. > If the calls still fail, Asterisk console output would be helpful. > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com/-- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > -- > Regards, > Mitul Limbani, > Chief Architech & Founder, > Enterux Solutions Pvt. Ltd. > 110 Reena Complex, Opp. Nathani Steel, > Vidyavihar (W), Mumbai - 400 086. India > http://www.enterux.com/ > http://www.entvoice.com/ > email: mi...@enterux.in > DID: +91-22-71967121 > Cell: +91-9820332422 > > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
