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support
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Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: May 14 2007 at 8:56am | IP Logged
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This post is for all LanScape VOIP Media Engine developers who are interested in integrating SAPI compliant TTS (speech synthesis) into their VOIP applications.
We have made available an updated sample application that will show you how to add TTS speech synthesis to your VOIP applications.
The SAPI 5.1 SDK ships with an example application called TtsApp. We have taken that sample application as the starting point and have added full VOIP Media Engine support.
The sample now has the ability to initiate and receive VOIP phone calls. It can auto answer incoming calls and it can also stream TTS generated speech to its single phone line.
Using this sample will help show developers how simple it is to take TTS generated sample blocks of audio data (TTS speech) and stream that sample block data to a “transmit IVR” phone line output.
The sample basically demonstrates what developers will have to do when creating TTS based auto attendant or IVR based TTS applications.
The sample will build using VC6, VS2003 or VS2005.
You can download the ZIP file image here:
http://www.lanscapecorp.com/DevResources/Media Engine Software Examples/VOIPTTSApp.zip
Thanks,
Support
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juice
Vetran
Joined: December 05 2006
Location: United States
Posts: 139
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| Posted: May 15 2007 at 11:51am | IP Logged
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This is very exciting news!
Unfortunately, the sample fails to compile with the LanScape engine we are using with the following errors:
OpenRxIvrChannel : function does not take 10 arguments dlgmain.cpp 1330
GetTxIvrSampleBlockSize : function does not take 5 arguments dlgmain.cpp 1870
And indeed, the functions it complains do not, as the error says, take that many arguments. What version of LanScape Media Engine is this demo compatible with?
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support
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Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: May 15 2007 at 12:06pm | IP Logged
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Hi juice,
You are correct. We failed to mention that this sample app was built against VOIP media engine v5.12.3.30.
Simply fix the offending lines to fit your media engine version. The API changes since v5.12.3.10 are limited to a few parameter additions in some of the APIs. You should have no problem getting it to build with your version.
Support
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sabdullah
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Joined: May 08 2007
Posts: 6
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| Posted: May 19 2007 at 6:30pm | IP Logged
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My provider has the following servers and I can't seem to get it to work witht eh tts example.
Which server do i use?
# proxy.jnctn.net
* provides "standard" SIP call delivery
* username/password used for termination and registration authentication
* originated calls delivered "by name" to multiple registered contacts
* will proxy calls to and from other domains (i.e. FWD)
* recommended for use with VOIP phones and ATAs
OR
# sip.jnctn.net
* provides for "trunked" SIP call delivery
* username/password used for termination and registration authentication
* originated calls delivered "by number" to a single registered contact
* recommended for use with SIP servers and PBXes
Username: duoserve
Auth Username: duoserve
VOIP Password: <password>
When i make an outbound call to a localhost where I have a softphone working,
your samples work. However, when I try to make a call through my voip provider junction networks,
it doesn't. What from the above information that describes my credentials with my voip do I have to fill in your configuration dialog
With the xlite softphone I put the phone number without a sip uri so in this case what do I put for the sip uri
sip:[apstnphonenumberhere]@[what uri goes here]
I tried sip.jnctn.net and proxy.jnctn.net
Tried all combinations of things and it still doesn't work. Can you give me exactly what we should fill out for configuration dialog and sip uri. I had no problems doing this with xlite softphone and it registers and calls fine from there so I am sure there is no weird things going on with the provider. I also wanted to know if you do c# custom development so we can get tts working from c# rather than cpp. Not to familiar with atl and its code is so intertwined with the code we actually need to wrap in c#. Have not touched atl since many years ago and don't have resources to allocate on extracting needed code. Thats why we wish there was more console apps with registrar and proxy related code so we can see clearly what code is telephony related and which is frontend. You guys know your code from a-z and for a newcomer deciphering code takes up a majority of the time since it is all over the place. We can figure it out but it will take some time to review all source and header files.
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support
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Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: May 21 2007 at 9:01am | IP Logged
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Hi sabdullah,
We would be glad to assist you in your configuration. If the xlite softphone can connect, then our VOIP Media Engine examples will connect also. It may be something as simple as using your WAN IP address instead of the private network IP address. We would have to look at the SIP session log to see what is actually happening.
If you want us to look into the required configuration, we will need two temporary user accounts (phone numbers/extensions) in the provider’s network. Make sure to email us the full authentication user name, password and realm for each user account.
We will post the configuration settings for the TTS app when we determine what is going on.
<<< You
I also wanted to know if you do c# custom development so we can get tts working from c# rather than cpp.
Support >>>
Yes we do custom software development for customers on a contract basis. Mainly using:
Native Code:
C/C++
Managed .NET Code:
C# and VB
<<< You
Thats why we wish there was more console apps with registrar and proxy related code so we can see clearly what code is telephony related and which is frontend.
Support >>>
We know exactly what you are saying. It makes perfect sense. In this case, please consider doing us a favor:
We have an area of the support forum where you can post your “wish list” product requests. It is located here:
Wish List - Features you need from our software
http://www.lanscapecorp.com/forum/default.asp?C=11
If you want to see a certain software example added to the media engine product, post a description of the new sample application you would like to see. Don’t forget to mentin what language (C/C++, C#.NET, VB.NET, etc). If you need a new feature in the media engine, post that too. The more information we get based on customer requests, the better we can serve you and all the others in the VOIP community. We look forward to your future “wish list” postings.
Repost if you want us to assist further,
Support
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LanScape Support Forum -> LanScape VOIP Media Engine™ - Technical Support
Topic: VOIP and SAPI TTS speech sample now available
