A Linux-Based Implementation of Mobility Using SIP

The session initiation protocol (SIP) is one of the most popular signaling protocols used in applications that require a session setup and parameter negotiation before actual communication, as well as for session management and termination. Though SIP was designed primarily for voice over IP (VoIP) telephony, it currently is used in a variety of applications, including those for multimedia conferencing and multimedia content distribution.

SIP allows two or more participants to establish a session consisting of multiple media streams using text-based request and response messages. A user, termed a SIP endpoint, is addressed by a SIP URL in the form of an e-mail address, such as sip:alice@vocal.com or sip:alice@192.168.0.2. The application used for communication is called the user agent (UA). Call initiation and modification is done through INVITE messages of SIP. Two endpoints can communicate with each other directly, or they can make use of a SIP entity called the redirect server. The user first sends the request for call initiation to this server, which queries a location service to retrieve the IP address and port of the other user. The location service keeps track of the current location of the users.

One of the ways to populate a location service is to use a SIP entity called the registrar. The registrar takes registration requests in REGISTER messages from users to keep a dynamic record of the current location of the user. In some situations, it is undesirable for a network to give away its users' actual IP addresses for security reasons. In this case, a SIP proxy server forwards SIP requests and replies to other SIP messages on behalf of the user. In the rest of this article, the user agent, the redirect server and the proxy server are referred to collectively as a SIP network. More details about SIP can be found in RFC 3261.

Mobility is a key aspect of many applications requiring signaling. A user using a wireless phone may move from network to network and still expect to be reached seamlessly by other callers, even while in the middle of a call. However, no open-source SIP implementation currently supports mobility. In this article, we provide the first open-source SIP implementation that supports mobility. We use the open-source VoIP software VOCAL from Vovida.org and add mobility support to it. The result is a complete, open-source SIP implementation with mobility support that other developers can use to build SIP-enabled mobile applications. The software supports seamless SIP signaling when the user moves from network to network, as well as automatic hand-off if the user is in the middle of a call while changing networks. The implementation is done on Linux and currently is in the process of being contributed back to Vovida.

VOCAL was chosen as the core of this work because this software suite is a robust implementation of the SIP protocol and its various entities and is used widely. It therefore was felt that adding mobility support to VOCAL would allow the software to be used by a wider community of mobile application developers. The potential to contribute back to the Open Source community and the chance to interact with an active SIP community also were notable motivators for us to opt for VOCAL.

VOCAL contains a SIP user agent and various types of SIP servers. The user agent is called GUA, which also can be run from a GUI front end called SIPset. The implementation of the SIP proxy server in this suite is called a marshal server (MS). The redirect server and the registrar in SIP are combined together as the redirect server (RS). The location service contains two types of information—a database of users' current locations and a database called the Dial-Plan. The Dial-Plan is used to keep track of users in other VOCAL systems; it is a mapping of SIP URLs to the appropriate SIP server's address to which the SIP request message should be forwarded by MS.

The communication between two independent VOCAL systems is achieved by one-to-one correspondence of a marshal gateway (MSGW) on each system. Thus, a VOCAL system needing to communicate with multiple other VOCAL systems requires multiple marshal gateways. A standard SIP network also requires the users to subscribe to that network for authentication purposes. This subscription is implemented in VOCAL in the form of provisioning to a server called provisioning server (PS). VOCAL systems are provisioned from a Java-based GUI. The data from PS is stored in the MS, which the latter uses to authenticate the user before accepting any message from the user, including a REGISTER message to register the user in the location service. VOCAL contains some other SIP entities that are not relevant to this work.