SIP programming for the Java developer

Deliver SIP-based services to Java applications with SIP Servlet

By Wei Chen, JavaWorld.com, 06/19/06

Session Initiation Protocol (SIP) is a control (signaling) protocol developed by the Internet Engineering Task Force (IETF) to manage interactive multimedia IP sessions including IP telephony, presence, and instant messaging. The SIP Servlet Specification (Java Specification Request 116), developed through the Java Community Process, provides a standard Java API programming model for delivering SIP-based services. Derived from the popular Java servlet architecture of Java Platform, Enterprise Edition (Java EE is Sun's new name for J2EE), SIP Servlet brings Internet application development capabilities to SIP solutions.

IT and telecom are converging. Network-IT applications, typically data oriented, are merging with communication applications. The increasing number of Call Me buttons appearing on Webpages is an example of this integration. The SIP Servlet Specification brings a familiar programming model to Java developers for building converged applications. This article gives a step-by-step introduction on how to use SIP Servlet to build a simple echo chat service.

Session Initiation Protocol

Defined in Request for Comments 3261, SIP is a protocol for establishing, modifying, and terminating multimedia IP communication sessions. Figure 1 is a simple example of using SIP to establish a VoIP (voice-over Internet Protocol) call:

All the white lines in Figure 1 represent the SIP communications. Caller sends a SIP INVITE request to invite the "callee" to establish a voice session. Callee first responds with a message that has a 180 status code to indicate the phone is ringing. As soon as phone is picked up, a response with a 200 status code is sent to the caller to accept the invitation. Caller confirms with an ACK message, and session is established. Once the session is established, the actual digitized voice conversation typically transmits via Realtime Transmission Protocol (RTP) with the session, as the red line in Figure 1 indicates. When the conversation ends, a SIP BYE request is sent, followed by a response with a 200 status code to confirm the session termination.

Here is an example of a SIP INVITE request and a response with a 200 OK status code:

 SIP INVITE request:
INVITE sip:callee@callee.com SIP/2.0
Via: SIP/2.0/UDP pc.caller.com;branch=z9hG4bK776asdhds
Max-Forwards: 70
To: Callee <sip:callee@callee.com>
From: Caller <sip:caler@caller.com>;tag=1928301774
Call-ID: a84b4c76e66710
CSeq: 314159 INVITE
Contact: <sip:caller@pc.caller.com>
Content-Type: application/sdp
Content-Length: 142


(content (SDP) is not shown)

SIP 200 OK response:

 SIP/2.0 200 OK
Via: SIP/2.0/UDP pc.caller.com;branch=z9hG4bK776asdhds;received=192.0.2.1
To: Callee <sip:callee@callee.com>;tag=a6c85cf
From: Caller <sip:caller@caller.com>;tag=1928301774
Call-ID: a84b4c76e66710
CSeq: 314159 INVITE
Contact: <sip:callee@workstation.callee.com>
Content-Type: application/sdp
Content-Length: 131


(content (SDP) is not shown)

As you can see, the format of SIP resembles HTTP. However, when compared to HTTP, SIP is:

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