What is SIP?
VoIP is an attractive option to simplify communication channels and minimise telephone costs.
SIP (Session Initiation Protocol) refers to a technology that is used to carry the signals that begin and end the streams of data essential to standard telephony, but send them by means of TCP/IP networks over the Internet.
The most common use of SIP is VoIP ? Voice over IP. With VoIP, telephone calls are not connected via the "classic" wired telephone network (PSTN), but instead are routed via the Internet protocol TCP/IP. Using a broadband Internet connection, phone calls can be made easily ? and even for free between VoIP users!
Other applications include Instant Messaging services and video transmissions. The SIP telephone logs on to the Internet telephone network, which then registers it as a connection device. This login is carried out automatically ? all incoming phone calls are then routed to the SIP telephone.
SIP is the most common signaling protocol used by current VoIP services. It's fair to say that it has become the de facto standard signaling protocol for VoIP. An alternative protocol is H.323, but this is much less widely used and is mainly of historical significance.
Although SIP only handles part of the functionality required for VoIP (i.e. the signaling, which is mainly for establishing and ending the connection), it is still of fundamental importance. When establishing a connection, for example, the devices negotiate via SIP which technology they will use for the actual language transmission.
SIP is the foundation for implementing telephony via Internet protocol ("Voice over IP ? VoIP"). It allows use of the Internet infrastructure for telephone calls. Ideally, the entire "classic" wired telephone network (PSTN) could be eliminated. Furthermore, phone calls between VoIP connections (for example between branch offices, home offices, etc.) can be made using just the Internet, which means these calls don't incur any phone charges at all.
From the wired telephone network (PSTN) SIP users can be reached via corresponding gateways ? and they can, in turn, make calls to PSTN users who don't yet have a VoIP connection.
These days, the voice quality with VoIP is so good that you may have received calls via SIP without even realizing it.
The following overview shows the bandwidth of the different Codecs:
G.729 ? ( 8 kbit/s / 31,2 kbit/s) best compression and lowest practical bit rate for data transfer.
G.711a bzw. G.711u ? (64 kbit/s / 87,2 kbit/s) normal Codec, with a higher data transfer rate than possible with G.729
GSM ? from ETSI (13 kbit/s)is on a par with mobile phone quality.
Free phone calls between all SIP telephone users!
The use of an Internet telephone is extremely simple and offers ? in addition to a free SIP account ? a regular phone number that you can be reached at from anywhere in the world. Your SIP account on the TerraSip web site receives incoming calls to your number and diverts them via the Internet to your Internet telephone.
Using TerraSip, you can make Internet calls with your SIP telephone for free (you only have to pay for your broadband Internet connection). That's why users of SIP telephones can all call each other for free!