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ftgman Intermediate
Joined: February 27 2005 Location: United Kingdom Posts: 9
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Posted: February 27 2005 at 9:34am | IP Logged
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Please bear with me as I am not very VOIP aware.
I have been tasked with finding development tools and information relating to a new product line for our company.
We currently supply groupware mail servers and would like to add VOIP functions to our system.
Could you let me know if your API (Media Engine) would allow the development of the following features, and would any additional components be required?
1. SoftPBX - We require to be able to route calls between LAN users and to VOIP call carriers?
2. TAPI interfacting. Can call handling be routed through a voice capable modem?
3. FAX. Can the API handle FAX calls? Just routing them, or actually detecting the data content?
4. Does the API allow development of answerphone type applications for each extension? Can the data be extracted as a WAV file to be stored in the users mailbox.
5. What is the maximum number of calls you could handle on a duel 3ghz p4 with 2gb ram, and do you have a licence available for this size.
I'm sorry if my questions are silly but I have only just started this task and found your site very quickly. I have read the forums and looked through the API but still need these things answered.
A list of API samples would also be nice. So that I can see what I will have to go on.
Thank you for any information you can supply and hopefully I will be buying a licence.
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support Administrator
Joined: January 26 2005 Location: United States Posts: 1666
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Posted: February 28 2005 at 10:09am | IP Logged
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Hi Richard,
Thank you for contacting LanScape Corporation. Because this forum area is for pre-sales related VOIP questions, we are happy to receive all posts regarding our products. We know that VOIP development/deployment can be challenging at times so there are no silly questions. :)
You asked quite a few questions so we will have to take them one at a time.
Item 1: SoftPBX
The requirements of software based PBXs are many. From your question however, we are assuming that you need to route VOIP calls from the network environment to the Public Switched Telephone Network (PSTN). This capability is often termed “PSTN VOIP Gateway”.
An application you develop around our VOIP Media Engine will allow you to do this. You will need appropriate PC compatible telephony hardware to interface to the PSTN side (hardware interfacing to ISDN, BRI, PRI, POTS, etc).
Here is an example of how this would work: Lest assume a person on the network side of your VOIP telephony system wants to call a PSTN number, they would make the VOIP call, the call gets connected from the user’s soft phone/IP phone/User Agent (UA) to your soft PBX (that uses the VOIP Media Engine). The soft PBX then looks at the destination information of the call when it arrives. The soft PBX determines that the call is for a PSTN number and not another VOIP network destination. The soft PBX communicates with its PSTN interface hardware and attempts to get an outside PSTN line. The soft PBX then dials out to the PSTN, answers the call on the network side, and then starts streaming media to/from the PSTN side to/from the VOIP network side. The caller on the network side hears exactly what a non-VOIP PSTN caller would hear. The soft PBX in this case acts as the PSTN Gateways to go from the network side to the switched PSTN side. It works the same in the other direction (PSTN incoming calls getting routed to VOIP network destinations). The LanScape VOIP Media Engine gives your applications the capabilities to perform tasks like those described above. However, it is the application developer’s responsibility to “connect all the plumbing”.
Item 2: TAPI Interfacing
The VOIP Media Engine has no knowledge of Microsoft’s TAPI interface. This is as it should be. The application you develop uses our VOIP Media Engine API to handle all the tasks associated with the VOIP network side. Your application could also use TAPI to communicate to any TAPI compliant telephony devices. As a matter of fact, in the example given above for the SoftPBX, your PSTN hardware suppliers may offer a TAPI interface to manage the PSTN telephony hardware you chose and to stream audio data to/from the device. There is no problem mixing the VOIP Media Engine API and TAPI together in the same application. If you want to use budget voice modems (like full duplex voice compatible WinModems), and TAPI allows you to stream audio to/from them, then call handling can be routed through the full duplex voice modem. Please be aware that there are telephony limitations in using generic voice modems for a PSTN interface. Issues such as lack of echo cancellation and far-end call termination detection are prevalent issues. We have done extensive testing regarding the use of voice modems for telephony interfaces and have found that for almost all applications, they should not be used. However, using a full duplex voice modem is possible and the data rate/format of 8k uLaw (64kbits per second) they generally use is natively supported by the VOIP Media Engine.
Item 3: FAX
Transmitting a digitized modulated FAX signal over VOIP can be accomplished if the VOIP endpoints involved exchange media (RTP VOIP traffic) using 8k uLaw sampled data. The VOIP media Engine supports this data format and rate and makes this possible. The VOIP media engine will not know that the media it carries is voice or modulated FAX. Remember, the VOIP Media Engine gives your application the ability to manage VOIP call session and media. What your application does with the media streams associated with a VOIP call is up to you. Note: The VOIP Media Engine can not synthesize the generation or reception of modulated FAX signals.
Item 4: Answerphone –WAV file support.
You>>>
Does the API allow development of answerphone type applications for each extension?
Yes.
Can the data be extracted as a WAV file to be stored in the users mailbox.
Yes.
Note: Our LanScape company voicemail system uses the VOIP Media Engine and performs similar actions you describe above. It’s basically an IVR type of auto-attendant, DTMF detection, voice record/playback etc. If you want to hear what it sounds like, go to the LanScape web site and give us a call from the “Contact Us” page or the “Sales and Marketing” page. Listen to the voice prompts and leave a message if you like.
Item 5: Maximum number of concurrent calls.
This is a good question. Unfortunately we do not currently publish a maximum capability of the VOIP Media Engine because how a customer uses the engine affects how it will perform. :( We understand why this question is asked by customers and are taking steps to develop basic call handling benchmark specs.
We currently license the VOIP Media Engine from 1 to 64 concurrent lines. This represents the line support most often requested by our customers. The largest number of lines we have ever licensed is 1024. The customer was developing a VOIP media bridge.
LanScape can license the VOIP Media Engine up to as many concurrent lines as required by the customer. For OEM pricing details, please contact LanScape sales staff. From a software standpoint, there is no licensing limit on the number of lines the VOIP Media Engine will support or a customer can purchase. As you stated, there is a performance limitation though.
Most of the demand placed on a machine has to do with handling the media streams associated with VOIP calls. The total number of concurrent calls your application will be able to handle depends on how you develop your software around our VOIP Media Engine. Because of this variable, we cannot give you a hard and fast answer. Our customers must perform their own final performance testing and scale appropriately. On a 3GHz P4 machine, you should be able to get 200-300 8k uLaw (64kbit/sec) concurrent calls per processor per 100 megabit ethernet NIC. It may be more. Your mileage will vary.
Item 6: A list of API samples
The VOIP Media Engine comes with example software applications (source code) you can build and try immediately. As of 2-28-05, the example apps that ship with the product are:
Multi-line Conference Server
Dual Line IVR Server
Telephony Echo Server
Dual line/multi-lined soft phone
Single line soft phone
This post intended to get your questions answered accurately. If you have further pre-sales questions, please do not hesitate to post back to this forum.
Best regards,
LanScape Support Staff
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support Administrator
Joined: January 26 2005 Location: United States Posts: 1666
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Posted: February 28 2005 at 10:39am | IP Logged
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Hi Richard,
We posted the ReadMe.txt file that is installed along with the VOIP Media Engine v5.10 software examples. You can find the new post at the following URL:
http://www.lanscapecorp.com/forum/forum_posts.asp?TID=30&PN= 1
This should give you the additional information you requested.
Thank You,
LanScape Support
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ftgman Intermediate
Joined: February 27 2005 Location: United Kingdom Posts: 9
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Posted: February 28 2005 at 10:57am | IP Logged
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Thank you for your extensive replies.
It looks like I found what I'm looking for. We will be placing an order shortly.
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