[RESOLVED] Media stream and SIPp

Asterisk Asterisk Support
1

Hi,

I had compiled SIPp 1.0 on FreeBSD 5.4. Asterisk 1.2.5 was installed with the samples on the server. I had conducted tests with various softphone, X-Lite, SJPhone etc, using extension 1000(the demo-congrats playback).

However, when using SIPp with the below command -

./sipp -sn uac -nr -l 1 -s 1000 -r 1 -i 192.168.80.137 -p 5060 -m 1 -mp 8000 -mi 192.168.80.137 192.168.80.53

the console output stops at -

Playing ‘demo-congrats’ (language ‘en’)

X-Lite was about to continue and finish the call as below -

– Executing WaitExten(“SIP/101-ea93”, “”) in new stack
– Timeout on SIP/101-ea93, going to ‘t’
– Executing Goto(“SIP/101-ea93”, “#|1”) in new stack
– Goto (default,#,1)
– Executing Playback(“SIP/101-ea93”, “demo-thanks”) in new stack
– Playing ‘demo-thanks’ (language ‘en’)
– Executing Hangup(“SIP/101-ea93”, “”) in new stack

Are there any options to simulate a complete call like any softphone/hardphone? Or is it because SIPp does not support media stream as a client?

Appreciate any advice.

Regards

2

Install the Sipp1.1 unstuble version there is a scenario named uac_pcap.xml that simulate a rtp media traffic.

3

Thanks. I got it working a few weeks back with RC5, also tested successfully with customised pcap streams(recorded with Ethereal). Will change title to RESOLVED.

4

Well… I try to test my Asterisk Pbx:
I put in extensions.conf
exten => 1000,1,answer()
exten => 1000,2,playback(fileaudio)
exten => 1000,3,Wait(1000)
exten => 1000,4,hung(up)

with sipp1.1
./sipp -sf uac_pcap.xml -s 1000 [IP_server_Asterisk]:5060 -r 5

My server can support 200 current call after it retrasmits some packet and some call fall down.

I used the RTP with “g711a.pcap” file and

m=audio [auto_media_port] RTP/AVP 8
a=rtpmap:8 PCMA/8000

Now i would like to simulate the same but with differet codec, ex: GSM

I don’t know if change the SDP in
m=audio [auto_media_port] RTP/AVP 8
a=rtpmap:8 GSM/8000
and doesn’t change the file*.pcap i can do my simulation.

In this way in the CLI in Asterisk with the command “show sip channels” i can see the request code GSM: result i can simulate max 200 call:

What did you do?
Do you think i should record a file *.pcac with gsm codec?

I am noty able to record it with ethereal, and advise?

How did you do?

5

Based on my understanding from your dialplan and sipp options, I assume that you are testing 2-way audio?

Need to know more as to what you meant by “some call fall down”, probably by posting the sipp result.

I had made changes to my dialplan, which was used to record inbound voice generated by the sipp command to Asterisk voicemail -

./sipp -sf uac_pcap_tt-monkeys_ilbc.xml -nr -l 40 -s vm -r 2 -i 192.168.80.140 -p 5078 -m 40 -mp 8888 -mi 192.168.80.140 192.168.80.140 -trace_err

As you would have noticed, the options were much less stringent than yours, but it attained positive outcome for my case -

------------------------------ Scenario Screen -------- [1-4]: Change Screen –
Call-rate(length) Port Total-time Total-calls Remote-host
2.0(0 ms)/1.000s 5078 39.29 s 40 192.168.80.140:5060(UDP)

Call limit reached (-m 40), 0.959 s period 18 ms scheduler resolution
0 concurrent calls (limit 40) Peak was 39 calls, after 19 s
0 out-of-call msg (discarded)
1 open sockets
21982 Total RTP pckts 0.064 last period RTP rate (kB/s)

                             Messages  Retrans   Timeout   Unexpected-Msg
  INVITE ---------->         40        0         0
     100 <----------         40        0                   0
     180 <----------         0         0                   0
     200 <---------- E-RTD   40        0                   0

     ACK ---------->         40        0
   Pause [   1000ms]         40                            0
          [ NOP ]
   Pause [  17000ms]         40                            0
          [ NOP ]
   Pause [   1000ms]         40                            0
     BYE ---------->         40        0         0
     200 <----------         40        0                   0

------------------------------ Test Terminated --------------------------------

----------------------------- Statistics Screen ------- [1-4]: Change Screen –
Start Time | 2006-04-05 01:06:12
Last Reset Time | 2006-04-05 01:06:51
Current Time | 2006-04-05 01:06:52
-------------------------±--------------------------±-------------------------
Counter Name | Periodic value | Cumulative value
-------------------------±--------------------------±-------------------------
Elapsed Time | 00:00:00:963 | 00:00:39:319
Call Rate | 0.000 cps | 1.017 cps
-------------------------±--------------------------±-------------------------
Incoming call created | 0 | 0
OutGoing call created | 0 | 40
Total Call created | | 40
Current Call | 0 |
-------------------------±--------------------------±-------------------------
Successful call | 3 | 40
Failed call | 0 | 0
-------------------------±--------------------------±-------------------------
Response Time | 00:00:00:000 | 00:00:00:028
Call Length | 00:00:19:346 | 00:00:19:262
------------------------------ Test Terminated --------------------------------

The “uac_pcap_tt-monkeys_ilbc.xml” file was modified from the original uac_pcap.xml, using a pre-recorded stream playing the Asterisk tt-monkeys.gsm audio coded with ilbc, at a rate of 2 calls/sec up to a max of 40 concurrent calls at any time. I played the recorded voicemails with Wavepad and the quality differences in comparison with tt-monkeys.gsm was acceptable.

Need more info on how you try capturing with Ethereal to help further. I had recorded 3 pcap streams, will email if needed.