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Re: [Sipp-users] SIPP rtp playing issues
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From: CT R. <ct....@gm...> - 2013-04-18 13:51:11
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I have the same issue with the following package SIPp v3.3-TLS-SCTP-PCAP, built Feb 15 2013, 06:11:36 The package can be found here http://pkgs.org/centos-6-rhel-6/epel-x86_64/sipp-3.3-1.el6.x86_64.rpm.html It seems the problem is that the RTP ports specified (even if only one call is made) are totally ignored. My command line is like this /usr/bin/sipp -sf client.xml -inf client.csv 192.168.14.217 -i 192.168.13.21 -p 4000 -mp 4010 -trace_msg -trace_err -trace_rtt -d 5000 -m 1 -r 1 The SDP in INVITE is like this: v=0 o=[field1] 53655765 2353687637 IN IP[local_ip_type] [local_ip] s=SIP Call c=IN IP[local_ip_type] [local_ip] t=0 0 m=audio [media_port] RTP/AVP 0 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 and the <nop> <action> <exec play_pcap_audio="pcap/g711a.pcap"/> </action> </nop> <pause miliseconds="40000" /> the g711a.pcap file is the one included with the sipp. A few versions ago it was all running well. Regards Tiberiu On 19.februarie.2013 23:49, Vijay Goje wrote: > I found the issue .RTP is sent on local interface whether I use -mi or not. > How can I fix this problem. > > Vijay. > > -----Original Message----- > From: Vijay Goje > Sent: Tuesday, February 19, 2013 3:28 PM > To: 'vijay kant gupta' > Subject: RE: [Sipp-users] SIPP rtp playing issues > > Tried removing the mi and mp parameter still not seeing rtp in the tcpdump/wireshark captures. > > Regards, > Vijay. > > -----Original Message----- > From: vijay kant gupta [mailto:vij...@gm...] > Sent: Tuesday, February 19, 2013 2:27 PM > To: Vijay Goje > Subject: Re: [Sipp-users] SIPP rtp playing issues > > remove mi and mp paraeter from client and server cli... > > On Wed, Feb 20, 2013 at 12:24 AM, Vijay Goje <vg...@ct...> wrote: >> I tried with same rtp port 5000 and didn't get any rtp packets. >> >> -----Original Message----- >> From: vijay kant gupta [mailto:vij...@gm...] >> Sent: Tuesday, February 19, 2013 1:37 PM >> To: Vijay Goje >> Subject: Re: [Sipp-users] SIPP rtp playing issues >> >> Just Do a quick check in 200ok response specify same rtp port 5000. >> >> and see whether u r getting rtp packets or not.. >> >> >> On Tue, Feb 19, 2013 at 11:14 PM, Vijay Goje <vg...@ct...> wrote: >>> Here is a trace where my sipserver is running on 10.10.20.15 and sip running on 10.10.20.17,where I see the sip traffic but don't see the rtp in the capture. >>> >>> Regards, >>> Vijay. >>> >>> -----Original Message----- >>> From: vijay kant gupta [mailto:vij...@gm...] >>> Sent: Tuesday, February 19, 2013 12:29 PM >>> To: Vijay Goje >>> Subject: Re: [Sipp-users] SIPP rtp playing issues >>> >>> Hi Vijay, >>> >>> Can you take trace at Bparty who recieve call and rtp. >>> >>> I want to see signalling as well as rtp... >>> >>> Regards >>> Vijay >>> >>> On Tue, Feb 19, 2013 at 10:54 PM, Vijay Goje <vg...@ct...> wrote: >>>> I am using the g711a pcap file that came with sipp. >>>> Attaching it for reference. >>>> Can anyone send me any other ulaw pcap file with rtp which is working. >>>> >>>> Regards, >>>> Vijay >>>> >>>> -----Original Message----- >>>> From: vijay kant gupta [mailto:vij...@gm...] >>>> Sent: Tuesday, February 19, 2013 12:20 PM >>>> To: Vijay Goje >>>> Cc: sip...@li... >>>> Subject: Re: [Sipp-users] SIPP rtp playing issues >>>> >>>> Can i have your pcap file... >>>> >>>> On Tue, Feb 19, 2013 at 10:19 PM, Vijay Goje <vg...@ct...> wrote: >>>>> Hi, >>>>> >>>>> >>>>> >>>>> I am trying to send rtp to a destination ipaddress using the >>>>> example in the sip documentation >>>>> >>>>> http://sipp.sourceforge.net/doc/reference.html#uac_with_media >>>>> >>>>> I see the total rtp packets sent but when I do a wireshark/tcpdump >>>>> capture I don't see the rtp in the capture >>>>> >>>>> ------------------------------ Scenario Screen -------- [1-9]: >>>>> Change Screen >>>>> -- >>>>> >>>>> Call-rate(length) Port Total-time Total-calls Remote-host >>>>> >>>>> 1.0(10000 ms)/1.000s 5060 9.01 s 9 10.10.20.17:5062(UDP) >>>>> >>>>> >>>>> >>>>> 1 new calls during 1.002 s period 0 ms scheduler resolution >>>>> >>>>> 9 calls (limit 10) Peak was 9 calls, after 9 s >>>>> >>>>> 1 Running, 10 Paused, 5 Woken up >>>>> >>>>> 0 dead call msg (discarded) 0 out-of-call msg (discarded) >>>>> >>>>> 3 open sockets >>>>> >>>>> 1178 Total RTP pckts sent 59.370 last period RTP rate (kB/s) >>>>> >>>>> >>>>> >>>>> Can anyone let me know what I am missing here? >>>>> >>>>> >>>>> >>>>> Here is how I am running the server >>>>> >>>>> >>>>> >>>>> sipp -sf uas.xml -p 5062 -i 10.10.20.17 -mi 10.10.20.17 -mp 7000 >>>>> >>>>> >>>>> >>>>> Here is how I am running the client >>>>> >>>>> sipp -sf uac_pcap.xml -d 10000 -s 0917100000 10.10.20.17:5062 -l 10 >>>>> -mi >>>>> 10.10.20.17 -i 10.10.20.17 -mp 6000 -r 1 >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> Regards, >>>>> >>>>> Vijay. >>>>> >>>>> >>>>> ------------------------------------------------------------------- >>>>> - >>>>> - >>>>> - >>>>> -------- Everyone hates slow websites. So do we. >>>>> Make your web apps faster with AppDynamics Download AppDynamics >>>>> Lite for free today: >>>>> http://p.sf.net/sfu/appdyn_d2d_feb >>>>> _______________________________________________ >>>>> Sipp-users mailing list >>>>> Sip...@li... >>>>> https://lists.sourceforge.net/lists/listinfo/sipp-users >>>>> >> > > > ------------------------------------------------------------------------------ > Everyone hates slow websites. So do we. > Make your web apps faster with AppDynamics > Download AppDynamics Lite for free today: > http://p.sf.net/sfu/appdyn_d2d_feb > _______________________________________________ > Sipp-users mailing list > Sip...@li... > https://lists.sourceforge.net/lists/listinfo/sipp-users |
