Monday, November 30, 2009


This article describes a RESTful SIP application server architecture.

Why do we need this?
SIP is the protocol of choice for Internet session initiation and control such as for VoIP or multimedia calls. Although SIP is similar to HTTP in many respects, there are crucial differences in the design. Two of the major difficulties among web developers in adopting SIP are (1) no existing SIP-based web tools similar to programming libraries for HTTP and XMPP on Flash Player, (2) the initial cost to get started with basic working system is huge with lot of specifications, e.g., for NAT and firewall traversal. On the other hand, web developers are used to building applications on top of HTTP which works for most cases out of the box. More recently RESTful architectures are gaining popularity among web services. In the absence of easy to use web tools for SIP and large set of specifications for a SIP system, web developers tend to resort to quick and dirty hacks which in the end are short term and not interoperable. Hence there is a need for a easy to use RESTful architecture for SIP-based systems that allows quick application development by web developers. This article proposes such an architecture.

What exactly is difficult?
SIP supports both UDP and TCP transports. Many earlier systems implemented UDP, whereas both transports are a must for SIP proxy servers. In client-server communication, with several clients behind NAT and firewall, UDP causes problem. Secondly, with UDP you also need the reliability of transactions and hence the transaction state machines in SIP. The SIP request forking and early media feature have created lot of stir and confusion among developers. Several other telephony-style features are also not needed for many Internet oriented SIP applications that do not talk to a phone network. The NAT and firewall traversal are defined outside core SIP, e.g., using rport, sip-outbound. A developer usually prefers to have an integrated application library and API that is quick and easy to use. Moreover with lots of RFCs related to SIP, it becomes difficult to figure out what specifications are core and what are optional for a particular use case. A number of new web-based video communication systems use proprietary technologies such as on Flash Player because of lack of a ready-to-use SIP library to satisfy the needs.

To solve the difficulties faced by web developers, a subset of the core features of SIP are needed as an easy to use API. Such an API could be available as a built-in browser feature or a plugin. Once the core set of resources are identified, rest of the API can be customized by the application server providers and developers, or in separate communities.

What use cases are considered?
SIP is designed to be used consistently in different use cases such as client-to-client communication, client-to-server as well as server-to-server. The core SIP says that each SIP user agent (application client) has both UAC (client) and UAS (server). In this article I refer to client as a user agent and server as an application server, which are different from SIP terminology. Since the target audience for the proposal is application developers, only the client-server interface needs to be considered. The backend application server can translate the client-server request to appropriate SIP messaging for server-to-server case if needed, e.g., for service provider's network you may need high performance UDP based server-to-server SIP messages.

What are the SIP-related resources?
Once we focus on a small subset of the problem -- define RESTful API for client-server communication to access a SIP application server -- rest of the solution falls in place naturally. In particular, the SIP application server will provide two core resources: "/login" and "/call" to represent list of currently logged in users and list of active calls. Additionally, it can provide user profiles of signed up users at "/user" which internally may contain things like voicemail resources for the user. The client uses standard HTTP requests, with some additional methods as shown below, to access the resources and interact with others. One difference with standard RESTful architecture is that the client-server connection may be long lived, and also used for notification from server to client. In that sense it does not remain pure RESTful.

Login: The SIP registration and unregistration are mapped to "/login/{email}" resource, e.g., "/login/". Doing a "POST /login/{email}" with message body containing your contacts, can be used to REGISTER. The response will return your unique identifier for the login resource, e.g., "/login/{email}/{contact_id}. Later, you can use "DELETE /login/{email}/{contact_id}" to un-REGISTER or a subsequent "PUT /login/{email}/{contact_id}" to do a REGISTER refresh. The actual representation of the login contact information can be in XML, JSON or plain text and is application dependent. For example one could combine the presence update including rich presence with the registration method. Clearly the login update requires appropriate authentication.
 POST /login/      -- new registration
request-body: {"contact": "sip:kundan@"}
response-body: {"url": "/login/", "id": 1, "expires": 3600}

PUT /login/ -- registration refresh
request-body: "sip:kundan@"

DELETE /login/ -- unregister

GET /login/ -- get list of contact locations
response-body: [{"id": 1, "contact": "sip:kundan@", ...},...]
Call: The call is split into two part: conference resource and invitation. The conference is represented using a "/call/{call_id}" resource, where a client can "POST /call" to create a new call identifier, or "POST /call/{call_id}" to join an existing call. The conference resource represents the list of participants in a call.
 POST /call             -- create a new call context
request-body: {"subject": "some discussion topic", ...}
response-body: {"id": "123", "url": "/call/123" }

POST /call/123 -- join a call
request-body: {"url": "/login/", "session": "rtsp://...", ...}
response-body: {"id": 2, "url": "/call/123/2", ...}

GET /call/123 -- get participant list and call info
response-body: {"subject": "some discussion topic",
"children": [{"url": "/call/123/2", "session": "rtsp://..."}]

Invite: Call invitation requires a new message such as "SEND". For example, "SEND /login/{email}" sends the given message body to the target logged in user. Similarly, "CANCEL /login/{email}/1" cancels a previously sent message it is not already sent. The message body gives additional details such as whether the message is a call invitation or an instant message. The message body is application dependent. The SIP application server does not need to understand the message body, as long as it can send a SEND message from one client to another. This makes a SEND more closer to an XMPP instead of a SIP INVITE. If the callee wants to accept the call invitation, it joins the particular session URL independently.
 SEND /login/     -- send call invitation
request-body: {"command": "invite", "url": "/call/123", "id": 567}

SEND /login/ -- cancel an invitation
request-body: {"command": "cancel", "url": "/call/123", "id": 567}

SEND /login/ -- sending a response
request-body: {"command": "reject", "url": "/call/123", "id": 567, "reason": ...}
Event: SIP includes an event subscription and notification mechanism which can be used in several applications including presence updates and conference membership updates. Similarly, one needs to define new mechanism to subscribe to any resource and get notification of a change. This gives rise to a concept known as active-resource. The idea is as follows: if a client does a GET on active resource, and does not terminate the connection, then the client keeps getting the initial state of the resource, as well as any future updates until the connection is terminated. The future updates may include the full state or a difference depending on the request parameter.
 GET /call/123          -- keep track of membership information
response 1: ... -- initial membership information
response 2: ... -- any addition or deletion in the membership

GET /login/ -- keep track of presence updates
response 1: ... -- initial presence information
response 2: ... -- subsequent presence updates.
Profile and messages: The SIP application server will host user profile at "/user/{user_id}". The concept of user identifier will be implementation dependent. In particular, the client could "POST /user" to create a new user account, and get the identifier in the response. It can then do a "GET /user/{user_id}" to know various URLs to get contact location of this user. It can then do a GET on that URL to fetch the contacts or do a SEND on that URL to send a message or call invitation.
 POST /user                            -- signup with a new account
request-body: {"email": "", ...}
response-body: {"id": "", "url": "/user/" }

POST /user/ -- send offline messages (voice/video mail)
request-body: {"url": "rtsp://..."}

GET /user/ -- retrieve list of messages
response-body: [{"url": "rtsp://...", ...]
Miscelleneous: There are several other design questions that are left unanswered in the above text. Most of these can be intuitively answered. For example, the HTTP authentication credential defines the sender of a message, i.e., SIP "From" header. The sequential or parallel forking is a decision left to the client application. The decision whether to use a SDP or XML-based session description is application and implementation dependent. For example, if the client is creating a conference on RTSP server, it will just send the RTSP URL in the call invitations. Similarly, for Flash Player conferencing it will send an RTMP URL in the call invitation. The call property such as participant's session description can be fetched by accessing the call resource on the server. Thus, whether an RTSP/RTMP server is used to host a conference or a multicast address is used is all client or application dependent. The application server will provide tools to allow such freedom.

Conclusion: A RESTful interface to SIP application server is an interesting idea described in this article. The idea looks feasible and doable using existing software and tools, and hopefully will benefit both the web developer and SIP community in getting wider usage of SIP systems. The goal is not to replace SIP, but to provide a new mechanism that allows web-centric applications to use services of a SIP application server and to allow building such easy to use SIP application server.

Several of the pieces described in this article are already implemented in Python, e.g., RESTful server tools, video conferencing application server, SIP-RTMP translation and SIP server and client library. The next step would be to combine these pieces to build a complete REST and SIP project. If you are interested in doing the project feel free to get in touch with me!


Henry said...

This approach is a promising bridge between an all out replacement of SIP 2.0 with a SIP3.0 = HTTP+REST. It could be used as a gateway function placed either in SIP servers or in SIP UAs.

Henry Sinnreich

Gustavo said...

Nice post, we were working in something similar some time ago.
The asynchronous notifications part could be improved with the support of "webhooks" and bidirectional HTTP in the medium term.

I also tend to think that we will end up replacing SIP with HTTP like Henry has commented.