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will
Junior
Joined: February 07 2005
Location: Canada
Posts: 48
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| Posted: March 03 2006 at 4:59pm | IP Logged
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Hi, support,
the senario is:
1. we set up a 56k modem based dialup network connection to simulate the narrow band network enviroment to test the g729 codec coming with the version 5.11 engine.
2. we run the softphone(it has been set to use G729 in defualt) based on v5.11 engine, the softphone could not do registration, and returned with registration timeout msg.
3. we run the same softphone as in 2. but unplugged the dialup connection, put the DSL back, then the softphone could do registration without any problem. we repeated 2. and 3. several times and also increased the timeout setting in the softphone to 30seconds, 1 minutes, 2 minutes, 3 minutes and 4 minutes, the results are same.
4. we started SjPhone with the dialup connection, it registered fine and could make calls with clear voice quality.
5. then we run the old softphone compiled upon version 5.10 engine with dialup connection, suprisingly, it did the registration fine, then we made calls, the audio connection is ok, but the quality is bad because there is no G729 in the old softphone, so this is normal.
6. all of the softphones we tested used same registration configuration, which means the server is same, user name is same, and the computer is also same.
so we are confused of this sympton, could you please help us to figure out what is going wrong,
thanks,
will
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will
Junior
Joined: February 07 2005
Location: Canada
Posts: 48
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| Posted: March 03 2006 at 5:03pm | IP Logged
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just to add sth to the question above, the softphone which was configured to use g729 had been tested under broadband enviroment to make sure it indeed be able to use g729
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: March 06 2006 at 8:33am | IP Logged
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Will,
We will look into it immediately. You have given a very good problem description. Good job.
We will repost another response shortly.
Support
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: March 06 2006 at 8:36am | IP Logged
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Will,
If you have user authentication enabled at your registration server, give us the user name and password you used for authentication.
Support
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support
Administrator
Joined: January 26 2005
Location: United States
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| Posted: March 06 2006 at 10:47am | IP Logged
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Will,
We performed dialup testing and everything looks ok. No problems registering over 33.6 kbps dialup connection. G729/G729A and iLBC (20Ms and 30Ms frame sizes) all function as expected. Normal full duplex unbroken voice was experienced.
Here is our setup:
RAS Server:
Windows XP Pro Sp1 using Conexant ACLink Modem. RAS connection configured to assign DHCP IP addresses to clients who call in. We used the LanScape Centrex Proxy server as our SIP and registrar proxy on the server machine. Tests were conducted with user authentication both on and off.
Client Machine:
Windows XP Pro Sp2 using Agere WinModem. We tied the modem phone lines of the 2 test machines together using a Viking POTS telco emulator.
We dialed into the server machine from the client. The client was authenticated and was assigned a DHCP IP address from the server.
We then started one of our utility test soft phones on the client machine that uses v5.11 of the VOIP Media Engine. After proper configuration, the soft phone registered with our server. We then called another utility test soft phone running on the server machine. Calls in both directions connect as expected and low bit-rate codec exchange functions properly.
What to do now?
Make absolutely sure that the soft phone you are using (and the VOIP Media Engine) is sending its REGISTER SIP message to the remote server using the dialup IP address of the server. Take a look at the SIP log that is generated by your test software to verify this.
If your test setup is drastically different from what we described above, then post to this forum your RAS client/server configuration including operating system versions and installed service packs.
Keep looking. It’s usually something very simple causing the grief.
Repost as required.
Support
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will
Junior
Joined: February 07 2005
Location: Canada
Posts: 48
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| Posted: March 08 2006 at 12:40pm | IP Logged
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Hi, support,
the problem has been solved, seems it was the network configuration problem, thanks to your help.
there are several other questions:
1. when using "SetAudioMediaFormat" to switch between codecs, we noticed the softphone was hang up automatically before the switching takes effect, does this mean we actually don't need to make the softphone hang up before calling "SetAudioMediaFormat"? the engine will take care of shutting down the phoneline by itself?
2.Is there any realtionship between "SetAudioOutDataType" and "SetAudioMediaFormat"? For example, "SetAudioOutDataType" in default sets the audio output type to AUDIO_BW_PCM_8k, when we call "SetAudioMediaFormat" to switch to AUDIO_BW_G729, shall we also call "SetAudioOutDataType" with AUDIO_BW_G729 to make them consistent? If yes, do we need to call "CloseAudioOutChannel" first before calling "SetAudioOutDataType" to switch to another audio output data type?
thanks,
will
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: March 08 2006 at 2:34pm | IP Logged
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Will,
1)
The SetAudioMediaFormat API procedure sets a single codec type that will be offerd when innitiating a call. It does not affect or terminate a call that is already in progress. Sorry but we are not exactly sure what you are asking here. Maybe repost some additional information regarding this question.
2)
You>>>
Is there any realtionship between "SetAudioOutDataType" and "SetAudioMediaFormat"?
<<<Support
None. SetAudioMediaFormat specifies the single codec that will be used for outgoing calls. The SetAudioOutDataType API proc specifies the data format/rate of sample blocks you can send to the internal "audio outout" playback mixer.
Repost as required,
Support
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will
Junior
Joined: February 07 2005
Location: Canada
Posts: 48
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| Posted: March 08 2006 at 4:14pm | IP Logged
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hi, support,
sorry, maybe we did not express clearly.
In our testing, when a call connection had been established and after we called "SetAudioMediaFormat" we noticed that the connection was shut down automatically and the softphone status went to "onhook", so that is why we wanted to know whether the engine shuts down the call connection automatically when it sees "SetAudioMediaFormat" is called. If yes, we don't need to shut down the connection everytime when calling "SetAudioMediaFormat", because the engine will take care it automatically.
Another question actually is: we noticed "SetAudioOutDataType" uses the same data type AUDIO_BANDWIDTH as "SetAudioMediaFormat" does, so we thought "SetAudioOutDataType" could also be bandwidth related. Then we wanted to know after we called "SetAudioMediaFormat" with AUDIO_BW_G729 as parameter, do we also need to call "SetAudioOutDataType" with AUDIO_BW_G729 too to make calls of those two apis consistent? because in default "SetAudioOutDataType" uses AUDIO_BW_PCM_8K.
if "SetAudioOutDataType" with AUDIO_BW_PCM_8K is always used regardless of current bandwidth, that means it is bandwidth independent, then we will just leave it there untact whatever how we call "SetAudioMediaFormat" to change the codecs.
if "SetAudioOutDataType" needs to be called to reset the audio output data type, do we need to call "CloseAudioOutChannel" to close the channel first then call "SetAudioOutDataType" to change the output data type?
regards,
will
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: March 09 2006 at 7:50am | IP Logged
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Will,
Thanks for elaborating more.
You>>>
In our testing, when a call connection had been established and after we called "SetAudioMediaFormat" we noticed that the connection was shut down automatically and the softphone status went to "onhook",
<<<Support
SetAudioMediaFormat will not terminate any calls that are in progress.
You>>>
Then we wanted to know after we called "SetAudioMediaFormat" with AUDIO_BW_G729 as parameter, do we also need to call "SetAudioOutDataType" with AUDIO_BW_G729 too...
<<<Support
You only need to call the SetAudioOutDataType API proc if you are using the audio output functions of the media engine. The SetAudioOutDataType proc has nothing to do with the codec selection of calls.
You>>>
if "SetAudioOutDataType" needs to be called to reset the audio output data type, do we need to call "CloseAudioOutChannel" to close the channel first then call "SetAudioOutDataType" to change the output data type?
<<<Support
Sure - you can do that. An alternate way would be to make sure any audio out data blocks you have queued are completely consumed by the audio output. To wait for all audio out blocks to be consumed, call the WaitForAudioOutComplete API procedure. Once that proc returns to your code, you can then call the SetAudioOutDataType proc to change to a new audio out data type.
These are all good questions. Repost as needed.
Support
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LanScape Support Forum -> LanScape VOIP Media Engine™ - Technical Support
Topic: please help with bandwidth(codecs) problem
