VoIP protocols and standards
VoIP endpoints typically use International Telecommunication Union (ITU) standard codecs, such as G.711, which is the standard for transmitting uncompressed packets, or G.729, which is the standard for compressed packets.
Many equipment vendors also use their own proprietary codecs. Voice quality may suffer when compression is used, but compression reduces bandwidth requirements. VoIP typically supports non-voice communications via the ITU T.38 protocol to send faxes over a VoIP or IP network in real-time.
Once the voice is encapsulated onto IP, it is typically transmitted with the Real-Time Transport Protocol (RTP) or through its encrypted variant, the Secure Real-Time Transport protocol. The Session Initiation Protocol (SIP) is most often used to signal that it is necessary to create, maintain and end calls.
Within enterprise or private networks, quality of service (QoS) is typically used to prioritize voice traffic over non-latency-sensitive applications to ensure acceptable voice quality.
Additional components of a typical VoIP system include the following: an IP PBX to manage user telephone numbers; devices; features and clients; gateways to connect networks and provide failover or local survivability in the event of a network outage; and session border controllers to provide security, call policy management and network connections.
A VoIP system can also include location-tracking databases for E911 — enhanced 911 — call routing and management platforms to collect call performance statistics for reactive, and proactive voice-quality management.