SIP connection

A SIP (Session Initiation Protocol) connection is a service offered by an ITSP (Internet Telephony Service Provider) that connects a company's PBX to the ordinary telephone system (PSTN) via Internet using the SIP VoIP standard. Using a SIP connection may simplify administration for the organisation as the SIP connection typically will use the same Internet connection that is used for normal data. This eliminates installation and maintenance of a BRI/PRI connection at each location. However, this almost always comes at the expense of the reliability of a real TDM PSTN connection.

Thanks to the SIP protocol, transmitting Caller ID and other call-related information is easy and even other information like pictures of the caller can be transmitted. Because SIP connections are digital, the voice is also transmitted digitally and can have excellent audio quality (see Codec) and issues from analog lines like connect and disconnect detection are simple. Also, there is no need to limit the telephone number to 10 digits. This makes it possible to assign Direct Inward Dialing numbers for every extension, so that the auto attendant in the PBX plays a less important role.

The challenge with SIP connections over the public Internet is call quality. If the traffic runs on the same connection with other traffic like Email or Web, voice and even signalling packets may be dropped and the voice stream can get interrupted. Businesses must take special preparations when they want to use SIP connections as their primary PSTN termination. Also, the general stability of the Internet connectivity becomes a critical issue. Because of this, many companies split voice and data up into two separate internet connections to solve this problem, so that the resource conflict on the Internet access side is avoided. Other devices perform traffic shaping in order to avoid this resource conflict, but they still depend on the merit of the service provider not to drop packets from the Internet to the PBX.

Registration is required if the end user has a dynamic IP address, if the provider does not support static hostnames, or if NAT is used. In order to share several DID numbers on the same registration, the IETF has defined additional headers (for example "P-Preferred-Identity", see RFC 3325). This avoids multiple registrations from one PBX to the same provider. Using this method the PBX can indicate what identity should be presented to the Called party and what identity should be used for authenticating the call. This feature is also useful when the PBX redirects an incoming call to a PSTN number, for example a cell phone, to preserve the original Caller ID.

The increasing concerns about security of calls that run over the public Internet has made SIP encryption more popular. Because VPN is not an option for most service providers, most service providers that offer secure SIP connections use TLS and SRTP for encrypting the traffic. The keys for SRTP are exchanged using RFC 4568 (SDES).

ee also

* IP Telephony

External links

* [http://www.sipconnect.info/ SIPconnect Technical Recommendation]