by Joe Gleinser
5. September 2009 01:01
If you're curious about SIP, you're not alone. SIP, Session Initiation Protocol, is a method for terminating dial tone or connecting handsets to a phone switch. It enables vendor competition on proprietary phone systems for the high-margin handsets sales. It also allows dial tone to be passed over internet connections at a higher capacity than a traditional PRI. Most major phone system manufacturers and software based systems support SIP trunking and endpoints. SIP is already in use as a means to transmit voice on many carrier backbones.
Here are several important SIP facts:
- SIP is run on a typical network connection. Like any VoIP you must have QoS end-to-end for consistently high quality service.
- Many providers offer SIP to anywhere by pushing it across the public internet. Be wary. Look for vendors who are bringing in their own managed internet connection.
- On 1 T1 you can have either 23 lines (PRI) or 45+ SIP trunks. If you have ethernet options your phone capacity can scale very easily.
- SIP supports much of the functionailty of the traditional PRI such as Direct Inward Dial and Caller ID.
- SIP handsets may not be a good deal. Most major VoIP manufacturers charge a per-handset license fee equal to about the cost savings. However in response to price competition vendors such as Avaya have released lower priced handsets such as the 1600 series.
- SIP handsets lack functionality available on the proprietary handsets
- SIP is implemented differently by many manufacturers. Ensure your manufacturer is supported by your provider. Not suprisingly Cisco and Avaya users have many options.
SIP is a mature product with significant cost advantages and flexibility. It deserves consideration as a trunking option for all new installs.