Finally an RFC for the SIP INFO message

January 15, 2011

SIP, like HTTP, is a request-response protocol. Each request has a method, that is standardized. A call setup uses the INVITE method, a call tear-down (hangup) use the BYE method. Those are simple to understand, as are a few of the other ones, like REFER for call transfers and MESSAGE for SMS-like text messages. The SIP INFO method has been an unspecified area used for left-overs from the SIP meal. A number of usages has been established. The new RFC – “Session Initiation Protocol (SIP) INFO Method and Package Framework” (RFC 6086) tries to clear up the kitchen and organize the dishes to continue the metafor. Read more

Asterisk: Distributed states using SIP

January 8, 2011

Asterisk was originally built as a stand-alone system, a single central point for all telephony communication. In short, a PBX. Nowadays, the Asterisk Open Source telephony server, is used in many ways – many of them not really being PBXs. All kinds of applications are being powered by Asterisk.

While building applications with Asterisk, you soon realize that you’re limited to that single server. It’s hard to scale and one limiting factor is that the call state is being held in one server. Many services depend on call states – if an agent in a call center is busy, you need to find an available agent. If a trunk to the PSTN is in use, you might want to find another way out. Call states are important.

Of course, the Asterisk project is now working on the long term solution, the Asterisk SCF and the applications that will be built using this framework. But that will take some time. Meanwhile, the Asterisk PBX team has been working on a few ways to distribute the call states between a group of servers. This article will be describing a few of the different architectures being worked on. Read more