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:vijaykant.it2...@gmail.com]
> 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...@ctigroup.com> wrote:
>> I tried with same rtp port 5000 and didn't get any rtp packets.
>>
>> -----Original Message-----
>> From: vijay kant gupta [mailto:vijaykant.it2...@gmail.com]
>> 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...@ctigroup.com> 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:vijaykant.it2...@gmail.com]
>>> 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...@ctigroup.com> 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:vijaykant.it2...@gmail.com]
>>>> Sent: Tuesday, February 19, 2013 12:20 PM
>>>> To: Vijay Goje
>>>> Cc: sipp-users@lists.sourceforge.net
>>>> 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...@ctigroup.com> 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
>>>>> Sipp-users@lists.sourceforge.net
>>>>> 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
> Sipp-users@lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/sipp-users
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