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A Communications Services Framework to fuel IP telephony deployment?
Posted by boneill42 on November 12, 2003 at 06:58 AM | Comments (10)
I've started the snowball rolling for a Communications Services Framework. (CSF)
IP telephony deployment has been slowed by some major hurdles, and we have yet to see the "enhanced services" that will really drive the market. I think a web services environment, hosted on Java.Net, through which enhanced communication capabilities could be delivered, combined with a client side API that allows developers to take advantage of these capabilities could solve the problems found in current IP telephony deployments and provide impetus for migration.
One of the first problems I'd like to tackle with the CSF is that of identity. Currently, there is no way to automagically discover a persons contact addresses given one of the set; SIP address, phone numbers, email address, AOL screenname, etc.
This causes many problems. If I am on a SIP phone, dialing a phone number, it will go out through a gateway to the publicly switched telephone network (PSTN). If the person I was dialing happened to be available via SIP as well, the phone call will still travel out to the PSTN, and back in through another gateway because there is no way to discover that the person I was trying to call is available via IP. There is no way to keep from traversing the PSTN. How inefficient? This needs to be remedied.
What if there were a web service, an Aggregate Directory Service (ADS), that given a users contact address, phone number for example, would return the set of contact addresses via which that user is available? It would solve the problem. A SIP Proxy Server could then utilize this web service to determine if it can keep the call on the IP network, before blindly passing it to the gateway. How beautiful? Lets build it.
Because this is not the only such service required by these next-generation communications networks, it makes sense to house the ADS in a larger more general framework. This is the Communications Services Framework.
Bingo. we have a plan. I just started working on the complete specification for the CSF and the Aggregate Directory Service. If you would ike to get involved PLEASE contact me.
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Comments
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Communications for the people
This is a great idea and if we are smart about it, we can leverage several of the other project within Java.Net such as the SIP Communicator, the SIP Proxy server, and several of the IM projects. Launching small with a capacity to grow big - upwards toward the billions, is key to keep the project growing. The CSF may grow to encompass other presenced based data besides humans, namely device and applications, Web services may be important to get a program running, but checking to make sure there is access to people and/or devices make the web service even more powerful.
Posted by: dtait on November 13, 2003 at 08:31 AM
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how to bypass PSTN when SIP UA active
Brian said:
One of the first problems I'd like to tackle with the CSF is that of identity. Currently, there is no way to automagically discover a persons contact addresses given one of the set; SIP address, phone numbers, email address, AOL screenname, etc.
This causes many problems. If I am on a SIP phone, dialing a phone number, it will go out through a gateway to the publicly switched telephone network (PSTN). If the person I was dialing happened to be available via SIP as well, the phone call will still travel out to the PSTN, and back in through another gateway because there is no way to discover that the person I was trying to call is available via IP. There is no way to keep from traversing the PSTN. How inefficient? This needs to be remedied.
My response:
It seems to me that the building blocks to address this problem already exist. If a SIP user enters an E.164 TN then ENUM can be used to map the number to one or more network addresses, including a SIP URL. The calling SIP UA can then initiate a call to this SIP URL by going through its SIP proxy. The SIP proxy contacts the SIP Proxy associated with the called party, which may complete the call itself or return a redirection proxy address (for the current serving proxy). This way the call can be completed entirely within the SIP domain without dumping to PSTN. If, however, the user is on the PSTN, then the redirection proxy may be a media gateway that will terminate the call to the PSTN.
Posted by: alicia on November 14, 2003 at 02:23 PM
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how to bypass PSTN when SIP UA active
Thanks for the comment, and I certainly agree, but the assumption in your scenerio is that the SIP user has the E.164 TN of the user they want to contact (or can derive it). What if the user only has an AOL screenname, but they want to send another user an instant message? It might eventually be delivered via SMS, but there is no way to get from the AOL screen name to a telephone number.
The framework I'm proposing is more generic. Simply; given a contact address, a set of contact addresses is returned. This doesn't tie the user or the service to any addressing scheme, or any network, and doesn't require any extra knowledge on the end-user's part.
Note, this does not preclude the use of ENUM, in fact it encourages it. From any contact address, a client will be able to discover a persons ENUM address, then use it to map to one or more addresses.
Posted by: boneill42 on November 14, 2003 at 06:46 PM
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JAIN SLEE for base
This looks like an interesting effort. I'm wondering if the JAIN SLEE API is part of the plan as the base on which to build this Communications Services Framework. After all SLEE stands for Service Logic Execution Environment, and it's a well thought out API that should definitely be leveraged. I am a little disappointed actually that SLEE hasn't experienced yet a significant level of interest from vendors, or open source developers. For instance, I know of no actual implementations of the API, open source or otherwise. Hopefully this project will see this API for its potentials, and spur the development of implementations, and applications built on top of it.
Posted by: mikeazzi on November 18, 2003 at 02:28 PM
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JAIN SLEE for base
Yes, JAIN SLEE is being considered and is currently our platform of choice. You are correct. It is the ideal environment in which to deploy such services. There are two implemenations of which I am aware; by OpenCloud and jNetX. It is my opinion that we are going to see more and more uptake of SLEE given the demands of real-time communication on the internet. IP Communication is going to be one of the strong motivators because it requires the high throughput and fault tolerance offered by the JAIN SLEE API.
Posted by: boneill42 on November 18, 2003 at 03:08 PM
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uploadable sip services.
Hi Brian!
Interesting idea. I'll be digging there. Do check out jain-sip-appserver.dev.java.net for our research project that allows untrusted users to upload SIP Services onto a server. It uses java bytecode re-writing and the jain-sp 1.1 API as a trusted computing base and java sandbox technology. It is still evolving rapidly but I think there is synergy here and we should collaborate. Also coming soon to a java.net project near you sip-meeting featuring the ability to set up relay meetings through an uploaded service.
Posted by: mranga on December 10, 2003 at 06:45 AM
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JAIN SLEE for base
jNETx (www.jnetx.com), has a commercial offer based on a Jain SLEE.
They have already a few operationnal applications in commercial networks.
Posted by: bolne66 on July 09, 2004 at 02:31 AM
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JAIN SLEE for base. There is now an open source implementation of JAIN SLEE, which will further promote the technology to a wider audience. See http://www.mobicents.org
Posted by: ivelin on May 24, 2005 at 12:27 PM
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