[SIPForum-discussion] redirect server
Chris Liu
chrisxliu at gmail.com
Wed Jul 25 20:03:22 UTC 2007
Here you go
Chris
[asterisk-dev] Re: SIPP Testing*Greg Boehnlein* damin at nacs.net
<asterisk-dev%40lists.digium.com?Subject=%5Basterisk-dev%5D%20Re%3A%20SIPP%20Testing&In-Reply-To=Pine.LNX.4.44.0606072001230.15012-100000%40nucleus.nacs.net>
*Wed Jun 7 21:48:13 MST 2006*
- Previous message: [asterisk-dev] SIPP Testing
<http://lists.digium.com/pipermail/asterisk-dev/2006-June/021162.html>
- Next message: [asterisk-dev] SIPP Testing
<http://lists.digium.com/pipermail/asterisk-dev/2006-June/021163.html>
- *Messages sorted by:* [ date
]<http://lists.digium.com/pipermail/asterisk-dev/2006-June/date.html#21166>
[
thread ]<http://lists.digium.com/pipermail/asterisk-dev/2006-June/thread.html#21166>
[
subject ]<http://lists.digium.com/pipermail/asterisk-dev/2006-June/subject.html#21166>
[
author ]<http://lists.digium.com/pipermail/asterisk-dev/2006-June/author.html#21166>
------------------------------
On Wed, 7 Jun 2006, Greg Boehnlein wrote:
>* Hello,*>* I am working on a bunch of issues, one of which is
trying to track *>* down proof that using CONFIG_ZAPTEL_MMX w/ a
Centos 4.3 2.6.9-34.0.1 *>* kernel causes FPU errors w/ various
codecs. This requires me to have a *>* high volume of SIP calls doing
lots of Ulaw to G729 transcoding over a *>* period of time, using a
modified g729 binary from Digium that logs when *>* the errors
occur.*>* In doing this testing, I have setup a box w/ the latest
SIPP *>* traffic generator (http://sipp.sourceforge.net) to make
boatloads of 120 *>* second calls to a Music On Hold extension that
simply plays Ulaw based *>* MOH to the caller.*
Just wanted to follow up on my post, archive it for the world and provide
some insight into what the issue was.
Asterisk's 1.2 RTP stack requires bidirectional traffic to actually send
traffic back. There was a patch in Mantis a while back that did
asynchronous RTP, but I was unable to find it. However, the fact that you
need to receive an RTP packet to SEND an RTP packet is the reason why no
RTP was originating from the Asterisk server.
So.. here is what I did to get things working:
1. Installed libpcap and libnet on my development box.
2. Built SIPP w/ Pcap Play suppor using "make pcapplay"
3. I grabbed a g729 RTP strean from a SNOM phone w/ TCPDUMP:
tcpdump -T rtp -vvv dst 207.166.192.106 -w g729.pcap
4. I then dumped the "uac_pcap" XML file and made some tweaks:
./sipp -sd uac_pcap > uac_pcap.xml
My edits consisted of modifying the following line:
<exec play_pcap="pcap/g711a.pcap"/>
To
<exec play_pcap="pcap/g729.pcap"/>
And, I changed:
a=rtpmap:0 PCMU/8000
To
a=rtpmap:18 G729/8000
5. I then executed the following:
./sipp -sf uac-g729.xml -d 10000 -s 451 gw4.n2net.net -l 96 -mi
207.166.192.254 -mp 5606 -i 207.166.192.254
And presto.. LOTS of calls and LOTS of RTP traffic. Once you send one RTP
packet to the Asterisk server, it responds back to the sipp RTP mirror
port, and things just chug along after that..
My results (not to promising)
------------------------------ Scenario Screen -------- [1-4]: Change
Screen --
Call-rate(length) Port Total-time Total-calls
Remote-host10.0(10000 ms)/1.000s 5060 877.47 s 2485
207.166.192.184:5060(UDP)
0 new calls during 1.000 s period 2 ms scheduler resolution
2 concurrent calls (limit 96) Peak was 96 calls, after 9 s
1501 out-of-call msg (discarded)
1 open sockets
6390121 Total RTP pckts 236.576 last period RTP rate
(kB/s)
Messages Retrans Timeout
Unexpected-Msg
INVITE ----------> 2485 11980 2301
100 <---------- 144 0 45
180 <---------- 0 0 0
183 <---------- 0 0 0
200 <---------- E-RTD 139 674 0
ACK ----------> 139 674
[ NOP ]
Pause [ 8000ms] 139 0
Var Pause [ 10000ms] 139 0
BYE ----------> 139 1209 130
200 <---------- 6 0 1
------- Waiting for active calls to end. Press [Ctrl-c] to force exit. --------
------------------------------ Scenario Screen -------- [1-4]: Change
Screen --
Call-rate(length) Port Total-time Total-calls
Remote-host10.0(10000 ms)/1.000s 5060 878.45 s 2485
207.166.192.184:5060(UDP)
0 new calls during 0.986 s period 1 ms scheduler resolution
0 concurrent calls (limit 96) Peak was 96 calls, after 9 s
1501 out-of-call msg (discarded)
1 open sockets
6397343 Total RTP pckts 234.380 last period RTP rate
(kB/s)
Messages Retrans Timeout
Unexpected-Msg
INVITE ----------> 2485 11980 2301
100 <---------- 144 0 45
180 <---------- 0 0 0
183 <---------- 0 0 0
200 <---------- E-RTD 139 674 0
ACK ----------> 139 674
[ NOP ]
Pause [ 8000ms] 139 0
Var Pause [ 10000ms] 139 0
BYE ----------> 139 1209 132
200 <---------- 6 0 1
------------------------------ Test Terminated --------------------------------
----------------------------- Statistics Screen ------- [1-4]: Change
Screen --
Start Time | 2006-06-08 00:21:38
Last Reset Time | 2006-06-08 00:36:15
Current Time | 2006-06-08 00:36:16
-------------------------+---------------------------+--------------------------
Counter Name | Periodic value | Cumulative value
-------------------------+---------------------------+--------------------------
Elapsed Time | 00:00:00:986 | 00:14:38:484
Call Rate | 0.000 cps | 2.829 cps
-------------------------+---------------------------+--------------------------
Incoming call created | 0 | 0
OutGoing call created | 0 | 2485
Total Call created | | 2485
Current Call | 0 |
-------------------------+---------------------------+--------------------------
Successful call | 0 | 6
Failed call | 2 | 2479
-------------------------+---------------------------+--------------------------
Response Time | 00:00:00:000 | 00:00:10:058
Call Length | 00:00:50:108 | 00:00:32:841
------------------------------ Test Terminated --------------------------------
2006-06-08 00:36:16: Aborting call on UDP retransmission timeout for
Call-ID '2429-17803 at 207.166.192.254
<http://lists.digium.com/mailman/listinfo/asterisk-dev>'.
sipp: There were more errors, enable -trace_err to log them.
[root at dmx-64
<http://lists.digium.com/mailman/listinfo/asterisk-dev>
sipp.cumulus.2006-03-20]#
--
Vice President of N2Net, a New Age Consulting Service, Inc. Company
http://www.n2net.net Where everything clicks into place!
KP-216-121-ST
------------------------------
- Previous message: [asterisk-dev] SIPP Testing
<http://lists.digium.com/pipermail/asterisk-dev/2006-June/021162.html>
- Next message: [asterisk-dev] SIPP Testing
<http://lists.digium.com/pipermail/asterisk-dev/2006-June/021163.html>
- *Messages sorted by:* [ date
]<http://lists.digium.com/pipermail/asterisk-dev/2006-June/date.html#21166>
[
thread ]<http://lists.digium.com/pipermail/asterisk-dev/2006-June/thread.html#21166>
[
subject ]<http://lists.digium.com/pipermail/asterisk-dev/2006-June/subject.html#21166>
[
author ]<http://lists.digium.com/pipermail/asterisk-dev/2006-June/author.html#21166>
------------------------------
More information about the asterisk-dev mailing
list<http://lists.digium.com/mailman/listinfo/asterisk-dev>
On 7/25/07, Avoseh Suru Albright <brightgold_79 at yahoo.co.uk> wrote:
>
> Hello Everyone,
>
> Could someone explain to me how to set up SER to work with SIPp, ?
>
> Regards
> Bright
>
> *Eugene Mednikov <em at HERMONLABS.com>* wrote:
>
> Greetings, Amit!
>
> That's the same as SER I assume. How would I configure it properly as
> Redirect server?
>
> Respectfully yours, Eugene Mednikov.
>
> Software Engineer
> Hermon Labs
> 972-4-6288001, 207
> 972-54-4520187
> em at hermonlabs.com
> www.hermonlabs.com
>
>
> ------------------------------
> *From:* amit [mailto:amit.v at pyronetworks.com]
> *Sent:* Tuesday, July 24, 2007 8:16 AM
> *To:* Eugene Mednikov
> *Cc:* discussion at sipforum.org
> *Subject:* Re: [SIPForum-discussion] redirect server
>
>
> Hi,
>
> Use Openser ProxyServer
>
> http://openser.org <http://openser.org/cgi-bin/mailman/listinfo/devel>
>
> Software Engineer
> Amit
>
>
> On Sun, 2007-07-22 at 21:06 +0200, Eugene Mednikov wrote:
>
> Greetings!
>
> We need some simple redirect server for testing. What may be used?
>
> Respectfully yours, Eugene Mednikov.
>
> Software Engineer
> Hermon Labs
> 972-4-6288001, 207
> 972-54-4520187
> em at hermonlabs.com
> www.hermonlabs.com
>
>
> _______________________________________________ This is the SIP Forum discussion mailing list TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion Post to the list at discussion at sipforum.org
>
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit
> http://sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
>
>
> ------------------------------
> Yahoo! Mail is the world's favourite email. Don't settle for less, sign up
> for your free account today<http://uk.rd.yahoo.com/evt=44106/*http://uk.docs.yahoo.com/mail/winter07.html>.
>
>
>
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit
> http://sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://sipforum.org/pipermail/discussion/attachments/20070725/6e5f1769/attachment-0002.html>
More information about the discussion
mailing list