Home > Java, SIP, Telecom > SIP stories, part 2: dialog forking

SIP stories, part 2: dialog forking

When I’ve changed a way my SIP stack deals with dialogs, I’ve encountered with a problem that two unit tests stopped working. These unit tests implemented “corner cases” which are not very well described in RFC 3261, so I implemented them based mostly on my own understanding of SIP. I had two options: to claim that those use cases are wrong and remove them, or to fix my implementation. Obviously, I’ve started with use case analisys.

When “200 OK” response is sent on INVITE, corresponding client and server transactions are instantly terminated on UAC, UAS and all proxies involved. UAS should retransmit “200 OK” until it will receive ACK, but ACK may go in a different path than INVITE. Since reliable delivery of “200 OK” response is specific for UA elements, it is impelemented not in transaction layer, but in UA core of TU layer.

Because of proxy forking, a sender of INVITE can receive dialog-establishing provisional responses from several UASes. When one of them answers with “200 OK” response, the proxy must cancel all other branches. However, some UAS may also respond with “200 OK” before receiving CANCEL. When this second “200 OK” response will arrive at UAC, a client transaction for INVITE will be already terminated. But this response should be delivered to upper layers. RFC 3261 specifies (in chapter that such responses should be matched against ongoing dialogs, and if no matching dialogs are found then new dialog must be constucted. Matching of a response against a dialog has a purpose: to cut off retransmissions of “200 OK” responses. A logic is simple: if response matches a dialog in confirmed state, then it should not be reported to application layer. However, all this idea about matching responses against dialogs has lots of flaws:

  • Described logic means that for any “stray” response UA must construct a dialog, then pass it to application layer. There are no means to check if response corresponds to request actually sent from this node. The idea is that only responses for sent requests should be processed, and processing should stop some time after receiving first “200 OK” response. But there are no means to ensure that.
  • Application is usually interested in context of a response. For example, it may be interested in knowing a request for which response was received. For normal responses a request could be obtained from client transaction. Descibed procedure doesn’t explain how this could be solved.
  • re-INVITEs are sent within existing dialogs. Retransmitted “200 OK” responses on re-INVITE will be always reported to application.

It is clear that RFC 3261 has a big flaw here. In my old implementation I had a workaround. Before sending an INVITE I’ve prepared a “dialog prototype”. All “stray” responses were checked against dialog prototypes by comparing “Call-ID” header, “From” header and URI of “To” header, and if there were a match then I’ve created a dialog based on a prototype. A prototype also held a reference on INVITE, so I could provide a full context for response. This homebrew solution worked, but I firmly decided to follow RFC 3261 to a letter. Searching for ideas I took a look at open source Java implementations of SIP stacks.

NIST implementation of JAIN SIP API matches “stray” responses against ongoing dialogs based only on RFC rules. If some early dialog matches a response, then initial transaction of this dialog is used as a context for response. If response doesn’t match existing dialog, this response is just passed to application layer. So, in this part NIST SIP stack is not compliant to RFC 3261. Some may argue that application can implement this functionality, but I don’t buy it: since there is no a context for response (request is unknown, transaction is null), an application can do very little.

Sailfin implements an interesting hack. They don’t actually terminate client transaction after receiving “200 OK” response. Instead, they transition a transaction into “established” state. This both helps to ensure that response is received for request which actually was sent, and also provides a context for a response at upper layers. Such approach seems to be a very good idea, except that it is not RFC 3261-compliant.

That’s all for today. As I promised to M.Ranganathan, I’ve pointed out problems with RFC 3261 and JAIN SIP API.  In upcoming article I’ll tell about second use case, how I’ve solved these problems, and why Sailfin’s hack is not a hack after all.

Categories: Java, SIP, Telecom Tags: , , ,
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