It is created in such a way so the SIP telephony clients can use UDP and TCP port 5060 to connect to the SIP servers and other SIP endpoints. The main purpose of SIP is to set up and tear down video or voice calls. In cases where initiation of the session is required, however, SIP is being used as well. Events Subscription, event notification and Terminal mobility are included there. Many SIP related RFCs in reality define the behavior for this solutions. All video/voice communications are made over separated session protocols, similar to RTP.
SIP protocol was made within IETF (Internet Engineering Task Force) – organization occupied with establishing Internet standards and general standards concerning TCP/IP protocols.
SIP-addressing is logical, of the same type as URL in HTTP. And the most natural – to use the electronic mail as the SIP address. There are permissible varied parameters by the address: for example, it is possible to indicate the regular city telephone number, additional telephone, and the parameters of modem connection etc.
Obviously SIP is considerably better than H.323 and complies with the understanding of IP-telephony as a global mass IP-service, and it should be implicitly considered right this way. And as SIP is able to serve not only telephony but also any communications at all in the real time over the IP-protocol; obviously it should become a base for other IP-services not yet existing but will appear in the future, as well.
A modern voice communication SIP protocol is widely used to make calls in the Internet. However, the final subscriber is mostly interested not in the very SIP standard but in the SIP telephony conception based on it. One one hand, all traditional conveniences for making telephone calls, compatibility with the international subscription base and the procedure of dialing numbers are kept.
Depiction of ‘’Locktek WP04 WiFi IP SIP Phone The Locktek WP04 WiFi IP SIP phone is a huge single boost in mobile communications. The device weights about 3oz, has a crystal clear color TFT (Thin Film Transistor) display and keeps standby mode up to 140 hour with a factual time on line at 7.5 hours spare. This SIP phone offers along with dimming screen and keypad feature, characteristics, that never were reached before by all previous SIP hardware, in sense of operational and stand by time. Phone has multi-dialing pad, when you can dial regular phone numbers, IP addresses and SIP accounts in a straight forward way. Two or more customers, handling two or variety WP04 SIP phones on local intranet, just can dial the IP of another WP04 device to get connection, in this case SIP account is not required to be involved.
Datasoft Networks is a full service VOIP provider. Datasoft prides itself in providing high quality origination and termination services both domestic and international routes. Among the many features of doing business with Datasoft are: multiple codecs aRead more