LanScape Support Forum: Brekeke SIP Inter-Op Issue Still Exist!

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Topic: Brekeke SIP Inter-Op Issue Still Exist!

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speedvoip
Vetran
Vetran

Joined: August 07 2008
Location: Canada
Posts: 156

Posted: March 27 2009 at 1:35pm \| IP Logged

Hi Randal:
With latest product image v6.0.0.11 and same testing on Brekeke SIP server, it is very pity that previous issue still exists even after making modification per what u instruct. SIP message log and server-side capture is still on our support account for your reviewing.

Thanks

George

support
Administrator

Joined: January 26 2005
Location: United States
Posts: 1666

Posted: March 27 2009 at 4:41pm \| IP Logged

Hi George,

I just looked at your SIP log you uploaded. I do not see anything in your transmitted INVITE request that should cause the INVITE that is sent to your Brekeke SIP server to be hair-pinned back to your soft phone. Very weird….. :(

I will be working again this weekend. I need to get to the bottom of this once and for all. There is something simple I am not seeing. We WILL get it resolved.

Please do the following for me: Send me an email describing the network IP:port of all of your VOIP elements (soft phone, Brekeke SIP server, VOIP service provider) and where they are deployed. I basically need the network description of how everything interacts.

Is it OK to call a PSTN number in the US using your VOIP service provider?

That way I can test outgoing calls to the PSTN using my own PSTN phone here in my lab.

Thanks,

Randal

speedvoip
Vetran
Vetran

Joined: August 07 2008
Location: Canada
Posts: 156

Posted: March 28 2009 at 3:36am \| IP Logged

Hi Randal:
Your required email has been send to u. Pls check it for troubleshooting. Any other help i can perform for u.
This evening I will be online for talking with u on issue. U can use that sip account to place call to my home, dialplan is 0118675583871818.

Thanks,

George

speedvoip
Vetran
Vetran

Joined: August 07 2008
Location: Canada
Posts: 156

Posted: March 28 2009 at 3:38am \| IP Logged

Hi Randal:
No problem. You can place any PSTN call to U.S with dialplan 1 + area code + phone number.

George

support
Administrator

Joined: January 26 2005
Location: United States
Posts: 1666

Posted: March 28 2009 at 10:12am \| IP Logged

Hi George,

Its 10:00 a.m. here. We are starting on this right now.

…don’t stay up too late tonight. :)

By the way – I did not receive the email:
Please do the following for me: Send me an email describing the network IP:port of all of your VOIP elements (soft phone, Brekeke SIP server, VOIP service provider) and where they are deployed. I basically need the network description of how everything interacts.

Also, please include descriptions if any VOIP element is located behind NAR router or firewalls.

Randal

Note:
For anyone who reads this post, George is located in Southeastern China and Randal is located in Minneapolis Minnesota – USA. A 14 hour time difference.

speedvoip
Vetran
Vetran

Joined: August 07 2008
Location: Canada
Posts: 156

Posted: March 28 2009 at 10:51am \| IP Logged

Hi Randal:
Strange for not receiving email. hmm... maybe it is due to our SMTP server failure. Any way now i am writing email to u again.
I will always keep waiting for your resolvement news and providing help on demand.

Thanks

George

speedvoip
Vetran
Vetran

Joined: August 07 2008
Location: Canada
Posts: 156

Posted: March 28 2009 at 11:21am \| IP Logged

Hi Randal:
Do u get email?

Thanks,

George

support
Administrator

Joined: January 26 2005
Location: United States
Posts: 1666

Posted: March 28 2009 at 11:26am \| IP Logged

Got it!

support
Administrator

Joined: January 26 2005
Location: United States
Posts: 1666

Posted: March 28 2009 at 10:03pm \| IP Logged

Hi George,

OK. You are not going to like what I have to tell you.

Remote testing at your location:
I spent the entire day Saturday remotely testing the Release 6 LanScape VOIP media engine against your trial Brekeke v2.2.6.2 SIP server.

I have no idea what is going on at your location, but your Brekeke SIP server seems to run buggy as hell… and that is a huge understatement.

When I started testing this morning against your server, all calls worked as expected. Not what I wanted. I wanted to be able to duplicate your test results so that I could track down a possible issue in the media engine.

I called out to the PSTN using my media engine based soft phone using extension 10001126253971101. I called a PSTN number here in my lab in the US. All calls connected without errors or problems.

I did notice an issue with receiving audio that was less than perfect but that is a totally different issue than having the media engine INVITE requests being hair-pinned back from your Brekeke server.

As I performed testing using your Brekeke SIP server, all of a sudden the calls started failing. I started to receive all sorts of 400 Sip errors in response to media engine transmitted INVITE requests. The only thing that seemed to clear these call errors was to re-REGISTER or restart the SIP server. Very strange.

I wanted to test all of the media engine supported codecs against your SIP server so I performed a series of test calls. While testing, all of a sudden, I too experienced the media engine’s INVITE request being hair-pinned back to my test soft phone!

I continued to experience the INVITE problem for a number of calls. I do not know what I did but eventually, your server started acting normal again. Not good.

I spend a long time with your server and your PSTN termination performing test calls and making sure the media engine was doing what it is supposed to do. For some reason, your server deployment seems buggy but I failed to isolate the culprit. Seeing that we (LanScape) have not used the Brekeke SIP server (formerly known as the OnDo SIP server) very much, I started to assume that the Brekeke server is a piece of garbage.

Not only did I test the Release 6 media engine against your deployment, I also tested the following SIP clients:

Phoner v2.03
Xten X-Pro v2
Conterpath eyebeam v1.5
Polycon PVX v8
SjPhone v1.6

All of these had problems too. None of them showed the INVITE hair-pinning but then again I did not test until I experienced that problem. All of these SIP clients started to receive SIP response 400 to calls. I don’t know why.

At this point I was not feeling too good about what you have deployed at your location.

Testing with an identical deployment at LanScape:
I decided to set up an identical scenario as you are trying to deploy. Media engine based soft phones all behind NAT routers. The Brekeke v2.2.6.2 SIP server depoyed in the global IP address space. I used our PSTN termination server here that is exposed to the global IP address space to emulate your PSTN service provider.

The Brekeke SIP server is pretty easy to set up. I created a single user account. Authentications are enabled for REGISTER and INVITE requests from all user agents (media engine soft phones). I also specified a single call routing rule that would take all outgoing calls and vector them to my PSTN gateway (IP address below is spoofed):

Code:

Call routing entry:

$request=^INVITE      $target=79.73.98.244

Everything works as expected. I put the media engine through its paces testing all supported codecs. Everything worked. Initiating calls, call on/off hold, canceling calls, etc. Not only that, all of the SIP clients in the above list also worked as expected!

As a side note, the Brekeke SIP server performed without an issue in my deployment. I was pretty impressed with how it handles the calls. It’s behavior is exactly what I would have expected. I saw no issues in my deployment compared to what you are having.

Where we go from here:
Something strange is going on at your location. I don’t know if your server machines have some sort of virus, a hacked operating system or something else going on. Maybe something is going on with your ISP in China that is messing with the SIP traffic. At this point, I am not sure what else can be done. I have verified your deployment enough here such that I could even recommend it to my customers. That is how sure I am that all is working. I hope I have not missed something.

We can discuss how we will handle additional support this next week.

Thanks George,

Randal

speedvoip
Vetran
Vetran

Joined: August 07 2008
Location: Canada
Posts: 156

Posted: March 29 2009 at 2:08am \| IP Logged

Hi Randal:
Wonderful! I think your experiment and testing procedure is described in detail and i also need to drill down to underlying behavior.

Any way, I contribue great thanks to u for efforts spent at weekend.

Ok, I will discuss support plan with u next week.

George

speedvoip
Vetran
Vetran

Joined: August 07 2008
Location: Canada
Posts: 156

Posted: March 30 2009 at 1:03pm \| IP Logged

Hi Randal:
Pls address me your sip log and packet capture at labs
for my analysis.

Thanks,

George

support
Administrator

Joined: January 26 2005
Location: United States
Posts: 1666

Posted: March 30 2009 at 7:37pm \| IP Logged

Hi George,

Hmmmm.... I'm not sure what you are asking. :(

Randal

speedvoip
Vetran
Vetran

Joined: August 07 2008
Location: Canada
Posts: 156

Posted: March 31 2009 at 9:50am \| IP Logged

Hi Randal:
You have mentioned u have identical setup of Brekeke at your labs and performed exact testing with media engine based softphone, so i believe u should have sip message log or packet capture. I want to give me for analysis and study on strange inter-op issue.

Thanks,

George

support
Administrator

Joined: January 26 2005
Location: United States
Posts: 1666

Posted: March 31 2009 at 2:58pm \| IP Logged

Hi George,

We placed Wireshark capture files and SIP logs for all VOIP elements in your support FTP account.

Please see the file:

“LanScape Deployment - Wireshark and SIP logs - Sent to customer.zip”

For a description of the deployment and call scenarios, see the “Deployment scenario.txt” document in the ZIP file.

Randal

speedvoip
Vetran
Vetran

Joined: August 07 2008
Location: Canada
Posts: 156

Posted: March 31 2009 at 10:54pm \| IP Logged

Hi Randal:
I got it. Perfect works!!

George

speedvoip
Vetran
Vetran

Joined: August 07 2008
Location: Canada
Posts: 156

Posted: April 01 2009 at 1:02pm \| IP Logged

Hi Randal:
I have compared your SIP message log with my own one as well as packet capture. It shows that your some INVITE requests include header Route whereas my own one has no this header. Why?

Thanks,

George

support
Administrator

Joined: January 26 2005
Location: United States
Posts: 1666

Posted: April 01 2009 at 3:32pm \| IP Logged

Hi George,

The “Route:” headers appear in secondary INVITE requests for call hold and un-hold operations. This is normal behavior for the media engine.

If you compare INVITE requests for the initial call setups, your SIP logs and my SIP logs should match.

Does this make sense?

Thanks,

Randal

speedvoip
Vetran
Vetran

Joined: August 07 2008
Location: Canada
Posts: 156

Posted: May 13 2009 at 10:48am \| IP Logged

Hi Randal:
I have found new hint on inter-op issue with Brekeke SIP server. There is trivial difference between my deployment and yours.

1. Authentication is disabled for Register and Invite, meanwhile no accounts are opened at brekeke.

2. Config Brekeke to work as "Thru" mode, which means that for any sip clients, their registrar is pointed to real SIP server such as VoipSwitch/Asterisk etc, whereas outbound proxy or proxy is pointed to brekeke.

So at above deployment, I have uploaded SIP client side and brekeke side packet capture file and brekeke sip server running log to FTP support account at /Brekeke SIP Server - InterOP directory for your analysis. (Note: I use original native singleline phone example as sip client)

I believe in u can find clues this time.

Thanks,

George

support
Administrator

Joined: January 26 2005
Location: United States
Posts: 1666

Posted: May 13 2009 at 5:21pm \| IP Logged

George,

Thanks for this post. When I get time, I will look further...

Randal

speedvoip
Vetran
Vetran

Joined: August 07 2008
Location: Canada
Posts: 156

Posted: May 19 2009 at 10:41am \| IP Logged

Hi Randal:
How is going with issue analysis? Hope u have some free time to look through it.

Thanks,

George

support
Administrator

Joined: January 26 2005
Location: United States
Posts: 1666

Posted: May 21 2009 at 11:47am \| IP Logged

Hi George,

Yes, we will be looking into this using your new information. Unfortiunately it will have to wait until next week. I know… Sorry for this. We are in the process of releasing v6.0.0.12 media engine and that has to be completed before we tackle this issue.

This weekend is the “Memorial Day” holiday here in the states. We will be back at work again this Tuesday.

Thanks George,

Randal

speedvoip
Vetran
Vetran

Joined: August 07 2008
Location: Canada
Posts: 156

Posted: May 21 2009 at 12:04pm \| IP Logged

Hi Randal:
Ok. NP! I hope to get official v6.0.0.12 release when it is available with many improvements and enhancements.

Thanks,

George

speedvoip
Vetran
Vetran

Joined: August 07 2008
Location: Canada
Posts: 156

Posted: May 28 2009 at 11:47pm \| IP Logged

Hi Randal:
How about issue tracking progress? v6.0.0.12 release is available?

Thanks,

George

speedvoip
Vetran
Vetran

Joined: August 07 2008
Location: Canada
Posts: 156

Posted: June 02 2009 at 3:50am \| IP Logged

Hi Randal:
Why not to give me feedback? Pls enlight me way.

Thanks,

George

support
Administrator

Joined: January 26 2005
Location: United States
Posts: 1666

Posted: June 02 2009 at 9:14am \| IP Logged

Hi George,

We are getting v6.0.0.12 ready for deployment. After that I will look into the issue you described. I plan on reserving time this week. We are very busy here – pleases forgive the delay.

I will post back to this thread when I have further info.

Best,

Randal

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