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Skype is the most common used VoIP software; it allows customer to make free through internet and also provide a cheaper alternate when calling fixed line telephone or mobile phone through public switched telephone network.
Skype uses Peer-to-Peer (P2P) to deliver voice or video data hence it can easily penetrate NAT and firewall. Skype uses proprietary encryption method to encrypt all voice and video data from one end to another. Four kinds of service can be provided by using P2P model: PC to PC; PC to Phone; Phone to PC and Voicemail. Three kinds of devices can be found in Skype network architecture: User-end PC; Super node; Authenticate Server. Skype client software installed on User-end PC listen to specific port to accept incoming calls and maintain a table that store all reachable super nodes and its addresses. User-end PCs satisfy certain eligibility criteria can act as super nodes, these criteria include amount of free memory, processor and internet connection speed. Skype Authenticate Server stores friend list corresponding to specific username and that information can be retrieved by login with correct username and password. When initiate a phone call, Skype on User-end PC will send out a UDP packet to "nearest" super node, if that packet blocked by NAT device Skype will use TCP to establish connection and use certain ports such as 80 and 443 to pass firewalls.
Compare to traditional fixed line telephones and mobile phones, Skype offers low cost phone calls and it became more and more popular for customer, on contrast some service providers make their effort try to block Skype and Rebtel in order to maintain their profit.
Protocols & Standards
H.323 was first proposed by International Telecommunication Union (ITU) in 1996 and was updated in 1998, it is a standard consist of many protocols include H.225, H.245, RTP and RTCP. Majority of early VoIP networks were built based on H.323. H.323 was designed to enable voice or video calls within local area network (LAN), but soon to be used by Internet Service Providers (ISP) use it to initiate and maintain calls over much large networks such as wide area network (WAN) or even Internet.
H.323 consists of four components: Terminals; Gateways; Gatekeepers and Multipoint Control Unit (MCU). Gateway was defined to be used between switched circuit network (SCN) and packet switching network (PSN), provides protocol conversion, media encoding and decoding. Gatekeepers to be used for network address translation, routing information exchange and access control between gateways. MCU enables voice or video calls between three or more end points, when more than three participates within one session setup by H.323, it is mandatory every participate establish a connection to MCU.
Maturity and inclusiveness are core advantages of H.323; Different manufacturers and ISPs are familiar with protocols used in H.323 hence it is easy for them to develop equipment that support H.323 standard. Conversely H.323 has limitations include scalability and integrity with public switched telephone network.
H.248 is defined by ITU and MGCP comes from Simple Gateway Control Protocol (SGCP). SGCP was developed by Cisco and Bell. MGCP and H.248 are different protocols but enjoys some similarity in protocol architecture. H.248 and MGCP cannot be classified as independent VoIP protocols. MGCP is a simple combination of gateway control protocol and IP device control standard. MGCP was replaced by H.248. With development of Next Generation Network, control protocol between media gateway controller and media gateway will be unified to H.248.
Session Initiation Protocol is defined by Internet Engineering Task Force (IETF); it is a signalling control protocol works on application layer which can initiate, modify and terminate multimedia communications include voice or video calls. Session participates can communicate by using multicast, unicast or hybrid of multicast and unicast. SIP not only support point to point voice or video communication but also support multi point voice or video conferencing.
Similar to Hypertext Transfer Protocol (HTTP) which are commonly used for web browsing and Simple Mail Transfer Protocol (SMTP) for email delivery, SIP can also be classified as a text-based protocol. SIP uses SIP-URI as unique address of other user where URI stands for uniform resource identifier (e.g. sip:username@host:port).
SIP does not have capability to work separately; Session Description Protocol (SDP) and Real-time Transport Protocol (RTP) are two protocols commonly work along with SIP. SDP can provide description to all devices participate in same session, these descriptions can be Session name and Intentions; Duration of Session; Media used in Session and Transport Address. RTP is capable of carry real-time data; it can encapsulate real-time data into data packets and forward them over IP network. SIP does not offer guaranteed service with Quality of Service (QoS); by contrast SIP has ability to corporate with Resource Reservation Protocol (RSVP) to reserve link resources for certain data flow. SIP can be used in conjunction with many transport layer protocols include UDP, TCP and RSVP mentioned above.
SIP User Agent, SIP Registrar Server, SIP Proxy Server and SIP Redirect Server are four component used to initiate SIP session. User Agents are devices that used by individual person such as personal computers, mobile phones and tablets. Registrar Server holds information contains location of User Agents. Proxy Server receives session initiate message from User Agents within its own domain, then forward that information to requested User Agent or another Proxy Server if that requested User Agent is in another domain.
Real-time Transport Protocol is an application layer protocol designed to handle media stream over Internet, proposed by IETF as RFC 1889.