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tomach
Intermediate
Joined: February 23 2007
Location: Poland
Posts: 22
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| Posted: March 01 2007 at 4:24am | IP Logged
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Hello!
Does ULaw and Alaw means that u support 711 codec?
I am waiting for .net support, but I have a quesition, have you ever tried your system in this scenario:
One computer with your application :
1. Is registering to sip proxy,
2. Is making 512 connections simultaneously connections would be to gateway to PSTN so codec 711 or 729 would be needed,
3. To each line (of those 512) different wave files are played.
4. After particular wave file has finished, subscriber has to choose DTMF numbers and your application has to accept it.
Have you tried it? Is it possible?
Best Regards,
Tom
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: March 01 2007 at 6:57am | IP Logged
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Hi Tom,
Thanks for your post.
<<< You
Does ULaw and Alaw means that u support 711 codec?
Support >>>
Yes – exactly.
<<< You
I am waiting for .net support…
Support >>>
That is good to hear. It should be available shortly. We are excited to get the “VOIP Media Engine .NET” support into the hands of our users. The buzz so far has been good.
<<< You
I have a quesition, have you ever tried your system in this scenario:
One computer with your application:
1. Is registering to sip proxy,
2. Is making 512 connections simultaneously connections would be to gateway to PSTN so codec 711 or 729 would be needed,
3. To each line (of those 512) different wave files are played.
4. After particular wave file has finished, subscriber has to choose DTMF numbers and your application has to accept it.
Support >>>
Yes – that will work but you must have a feeling for the limitations that may exist.
Assuming you are running your VOIP server app on a fast host PC the app shouldn’t be the issue. It’s too bad but at the moment we do not publish media engine performance specs due to the huge number of variables involved. For example: Host PC speed, network hardware throughput, the architecture and design of your VOIP application software that wraps the media engine, etc… We are working to get some sort of “normalized” performance benchmarks published that will help users with deployment decisions. More to come on this later…
The bandwidth of your network connection will probably be the limiting issue. Roughly speaking, if you run 512 simultaneous lines using G711 (aLaw or uLaw) codec, the rough bandwidth required would be (512 * 8k bytes/second) which is about 4 megabytes per second over the network. Is your network connection that fast?
If you use the G729/G729A codec support in the media engine, then your rough network bandwidth requirements can be reduced by a factor of 8 which would be 0.5 megabytes per second. Also, with that many lines running in your app, use G729A. It takes ½ of the computational power of G729 but for all practical purposes you get the same voice quality and is bit compatible with G729.
Remember: The rough numbers we stated above represent RTP media traffic only. There will be additional network burden associated with the SIP call setups and teardowns. The impact of SIP relative to the demands of RTP media however are relatively small in comparison. Also what is not taken into account in the above rough numbers is RTP header length which is 12 bytes per media packet at 50Hz per channel regardless of codec used and UDP and IP headers.
Notes regarding DTMF decoding:
If you use G711 codec, no problem. The media engine can decode the in band DTMF digits for you if that’s what you want. Otherwise you can have your app monitor incoming RTP media packets for out of band RFC compliant DTMF payload packets and process them accordingly.
If you use G729/G729A, then your app will have to monitor incoming RTP media packets for out of band RFC compliant DTMF payload packets and process them. The processing of out of band DFMF will be added to an upcoming release of the media engine. Having your app process the out of band DTMF is not hard and we have apps here that do it. We just have to move that functionality into the media engine to make life easier for users.
We hope this information has helped to answer your questions. If not, please repost so we can go further.
Good questions.
Support
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tomach
Intermediate
Joined: February 23 2007
Location: Poland
Posts: 22
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| Posted: March 02 2007 at 4:00am | IP Logged
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Hello!
Thanks for long good answer.
So if I understand right you are going to add detection of DTMF soon? (in g729).
I tried your application, to check voice quality and so on...with g.729.
I tried MultiLine Phone, but when i try to call it asks me for adressbook.txt?
Can you tell me how syntax of this addressbook should look like and where I should put it?
In this application I can use two lines, is it possible to try it on more then two with this application?
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: March 02 2007 at 6:52am | IP Logged
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Hi Tom,
<<< You
So if I understand right you are going to add detection of DTMF soon? (in g729).
Support >>>
Just to be clear: The media engine can already perform in band DTMF detection. What it natively does not yet support is out of band DTMF detection. This however is not a real huge issue seeing that applications have access to all raw RTP media packets on a per phone line basis and can filter/detect out of band DTMF RTP packets itself. We do it this way right now for certain customer custom applications we have been contracted to develop.
Adding native support for out of band DTMF would be introduced after this next upcoming v5.13 release which is focused mainly on maintenance issues, some enhancements and .NET manages code support. We are working as fast as we can to add this new DTMF capability but there is no definite time frame when it will be released.
<<< You
Can you tell me how syntax of this addressbook should look like and where I should put it?
Support >>>
The address book for the sample apps simply contains SIP URIs to call. Here is an example of the file:
Code:
;Start_Address_Book
;SIP URIs:
;Note: Each line in the address book is a SIP URI.
;For example: sip:1234@Mycompany.com:5060
;----------------------------------------------------------- ----------
"sip:900@lanscapecorp.dnsalias.com:6000"
"sip:901@lanscapecorp.dnsalias.com:6000"
"sip:902@lanscapecorp.dnsalias.com:6000"
"sip:1234@Mycompany.com:5060"
;End_Address_Book
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You should have a default address book like the one above in your install directory. For example:
C:\Program Files\LanScape\ VOIP Media Engine\5.12\Software Examples\Bin\Debug
or
C:\Program Files\LanScape\ VOIP Media Engine\5.12\Software Examples\Bin\Release
Look for AddressBook.txt in one of those directories.
Make sure the AddressBook.txt file is located in the same directory as the example app you are trying to run.
If you wanted to create a new default address book for the sample apps, this is the minimum info you would need in the file:
Code:
;Start_Address_Book
"sip:1234@YourCompany.com:5060"
;End_Address_Book
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<<< You
In this application I can use two lines, is it possible to try it on more then two with this application?
Support >>
Yes but you will have to modify the sample source code to do it.
Thanks Tom. Repost as needed,
Support
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tomach
Intermediate
Joined: February 23 2007
Location: Poland
Posts: 22
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| Posted: March 08 2007 at 4:55am | IP Logged
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I have following question. I am lost. Registration seems to be so easy usually but in this case I can not do it :( I would like to register to sip proxy. But I have tried to use all options in configure ant can not register :(
I've got this kind of parameters to register:
proxy sip: sip.test1.com
login: 300
password: bbb
Its our local company sip proxy.
Can you advice me which option should i tick and where to put all those infomation to register?
I tried all combination and it didnt work while with other softphones it works very good.
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: March 08 2007 at 6:51am | IP Logged
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Hi Tom,
What SIP proxy are you using?
Are you using one of the software examples that come with the media engine?
For example, if you are using the single line soft phone sample, here are the basic settings you will need:
--- Sip Proxy and Register Servers ---
Use Sip Domain:
Checked
Sip Domain Name:
Whatever your sip proxy domain name is (sip.test1.com)
Use registration server:
Checked
Server address:
The host name or IP address of your sip proxy or co-located sip registrar server.
Registration interval:
Set to 3600 for 1 hour.
Server port:
Set this to the sip proxy port or the co-located sip registrar server.
Time out Ms:
Set to a nominal value of 4000 Ms.
Use sip proxy:
Checked
Sip proxy address:
The host name or IP address of the SIP proxy.
Sip proxy port:
Usually 5060 however set this to the SIP server port your sip proxy is using.
Wan IP:
Not checked.
Send sip keep-alive packets to proxy:
Not checked.
You only need these next settings if your SIP proxy has authentication enabled:
--- Challenge Authentication ---
Use Authentication:
Checked.
Realms:
Usually you specify your SIP domain name here. If your authentication realm on your SIP proxy is not the same as your sip domain name, then specify your sip proxy authentication realm.
Login names:
Specify a valid authentication login/user name.
Passwords:
Specify a valid authentication password for the above specified login/user name.
That should be it. If you cannot register with your sip proxy, we need to see a SIP log. It could be a possible bug or most likely a configuration problem. Easy to determine once we see a sip log.
To make then single line soft phone create a sip log file, you will have to rebuild the app. Open the Phonebase.cpp file. Somewhere around line 1470, you will see the following 2 lines of code:
Code:
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StartupParams.LogSipMessages = FALSE;
StartupParams.pSipLogFileName = "SipMessageLog.log";
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Here is what you can do:
1)
Change the line:
StartupParams.LogSipMessages = FALSE;
to:
StartupParams.LogSipMessages = TRUE;
2)
Rebuild the app.
3)
Run the rebuilt app and let it try to register with your sip proxy.
4)
Exit the app.
5)
Post the sip log file info from the "SipMessageLog.log" file to this forum thread and we take a look to locate the problem.
Note: When you rebuild the sample apps, make sure you use the LanScapeVME.C source file that came with your license. You should place the LanScapeVME.C file in this directory:
C:\Program Files\LanScape\VOIP Media Engine\5.12\Software Examples\Microcode
There may be a bogus LanScapeVME.C file already in that directory. In this case just overwrite the default file that is there (the bogus file contains all NULL data).
Repost as needed,
Support
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tomach
Intermediate
Joined: February 23 2007
Location: Poland
Posts: 22
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| Posted: March 09 2007 at 8:36am | IP Logged
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Hello!
Multi phone works weird. Maybe it is because of trial version but it disconnect me sometimes after 5 seconds :( after I pick up the phone. Next I can not make two connections simultaneusly. I mean if I make second the first one hold and vice versa. So when I looked at the channels of my media gateway, only one was taken. Is it possible that two were taken?
I guess it is about implementatnio of your demo application but i do not feel like implementing in unmannaged code so Ill wait for your .NET solution.
Would it be dll?
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: March 09 2007 at 11:53am | IP Logged
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Hi tomach,
<<< You
Multi phone works weird. Maybe it is because of trial version but it disconnect me sometimes after 5 seconds.
Support >>>
We would have to see a SIP log of the call to determine what is going on. Is your trial expired?
<<< You
Next I can not make two connections simultaneously. I mean if I make second the first one hold and vice versa.
Support >>>
Yes. That’s the way the dual line phone sample was created. You can do what you want if you change 1 line of code. Look at the MultiLine PhoneDlg.cpp module. Locate the BOOL CMultiLinePhoneDlg::InitializePhone(void) member function. The member function calls the StartTelephonyEngine() procedure. Change the call to look like this:
Code:
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//StartTelephonyEngine(2,PHONE_LINE, TRUE,TRUE);
StartTelephonyEngine(2,SWITCH_LINE,T RUE,TRUE);
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You should now be able to conduct 2 simultaneous calls without the media engine performing automatic call hold logic.
<<< You
So when I looked at the channels of my media gateway, only one was taken. Is it possible that two were taken?
Support >>>
When you have 2 calls active at the same time with the sample phone (even though 1 of the calls is on hold), I would think that your media gateway would show 2 RTP media sessions active.
<<< You
…so Ill wait for your .NET solution. Would it be dll?
Support >>>
Yes. The .NET support will be a managed DLL wrapper/assembly. You can use in your managed code projects by adding a reference to the LanScape media engine managed DLL.
Note:
This thread is getting pretty long. Feel free to post additional questions to a new thread.
Repost as needed,
Support
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LanScape Support Forum -> LanScape VOIP Media Engine™ - Pre-Sales Technical Support
Topic: Important question about Media Engine - Deployment
