JTAPI with CTI ports to integrate with Cisco Unified Call Manager version 8.5

Question

What is JTAPI and what is CTI and what is Cisco unified call manager ? How can i integrate a sip call or voip call with this case?

Answer 1

This is a very open ended wide question, and you will have lots of basic research ahead of you before you are ready to run with this. To get you started: CTI Stands for "Computer Telephony Integration", and is a generic term for controlling or monitoring Telephony systems, such as PBXs, using an API. CTI Descrption at Wikipedia. Cisco Unified Call Manager is what Cisco is calling their PBX offering these days. Basically its a box that looks a bit like a router, you can connect VoIP based phone sets to it via an ethernet port and it speaks SIP or Skinny (Cisco's proprietary signaling protocol) as well as RTP to the phones. CUCM is the core component of a business's telephony system if they are using Cisco technologies.

There are two basic flavors of CTI- First Party Call Control, and Third Party call Control. First Party is where your application acts like or is associated with a single phone extension and can sent commands to get information about and control what that phone is doing. For example, your program could tell if a phone is ringing, take the phone offhook, dial a number, etc. This is often used for things like Address book integrations, where the user can use a program to click a contact on the screen and have their phone dial that contact. The two most common protocols for First Party Call Control are TAPI and JTAPI. (JTAPI also supports some 3rd Party call control features, though a lot of systems that provide a JTAPI API don't provide these)

Cisco CUCM provifes both TAPI and JTAPI, though I've often heard of their TAPI implementation described as quirky and undesirable, so JTAPI is more often used. JTAPI provides an API for interacting with the telephony system from a Java program.JTAPI Description and Links. A more comprehensive API for 3rd Party Call control is TSAPI, but Cisco CUCM does not provide a TSAPI style API unless you also have the optional Unified Contact Center module which is expensive, and even then the API is pretty rough (basically just a spec for binary IO over a socket and an API that provides the TSAPI style call state model).

As for how to integrate a SIP or Voip Call using JTAPI, it depends on what you wish to do. For example, it is possible to use a feature called 'Built in Bridge' on the latest generation of Cisco phonesets, to provide a SIP Trunk terminating at your program and to send a SIP transmission of every call going to/from each phoneset or a subset of calls controlled by you to your program in real time. This way you receive 2 RTP feeds for each conversation-- the local end, and the far end. THis is useful for passive applications such as monitoring or recording.

For more active applications, you can use Cisco's JTAPI and other APIs to remotely monitor or even control a physical IP Phone or Softphone on the system, or even register your application as a Virtual extension and take calls, join conferences, force transfers of calls to third parties, etc.

For more information see: http://developer.cisco.com/web/jtapi

Answer 2

I believe Your query about CTI and JTAPI are answered pretty well by "bdk" , let me answer you query about SIP integration of call manager in VoIP deployments , with a different perspective not involving CTI or JTAPI.

To begin with SIP ( Session initiation protocol ) is used to establish VoIP connections across a network.

please refer to this link for more detailed discussion on SIP : http://www.ietf.org/rfc/rfc3261.txt

Another Very Good Document talking about SIP in CUCM environment is : http://www.ciscopress.com/articles/article.asp?p=664148&seqNum=2

You have phones which are sold by Cisco , operation on the SIP protocol there are many Third party SIP phones in the market as well which can be "registered" to a call manager.

When these Phones "Register" to a particular call manager they talk to the call manger through SIP protocol meaning the "signaling between the call manager and the phone is SIP".

SIP is also used in many other flavor For example in SIP trunks to other gateways and in CME-UCCX integration .