You need to compile sipp with pcap support. Here is an example scenario: http://sipp.sourceforge.net/doc3.0/reference.html#UAC+with+media On Sep 22, 2009, at 5:13 AM, DHAVAL INDRODIYA wrote:
> Hello > > I would like to play file with sipp command. > > I want to take value of RTPAUDIOQOS for every user.. I will make it > hard testing with 500 users. > > But when all user leave from this conference I am unable to receive > proper value for highlighted in below line.. > > ssrc > = > 877077954 > ;themssrc > = > 0 > ;lp > = > 0 > ;rxjitter > =0.000000;rxcount=0;txjitter=0.000000;txcount=0;rlp=0;rtt=0.000000 > > However, when i made single call using SIP phone then i will receive > all value from RTPAUDIOQOs. > > Any Idea.. how can I play or transfer/receive Audio packets while > testing with SIPP command [using below command] > > I need specially value of receive streams . > > ./sipp -sn uac -d 10800000 -s 8601 127.0.0.1 -l 50 -r 1 -rp 5000 > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 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 _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
