Voice over IP or Voip service uses the Internet Protocol (IP) to transmit both voice and data communications in form of packets over an IP network. With Voip protocols, voice communications can be accomplished on whichever IP network being used – whether it is Internet, Intranets or Local Area Networks (LAN).
A Voip service digitizes, compresses and converts voice signal into IP packets and then transmits them over the IP network. The Voip protocols help in setting up and ending calls, plus help in delivering required information to other users. The fundamental benefits of Internet phone or Voip phone include the cheapest calls rates, all the while with using internet, fax and voice mail through the integration of data, voice and video on one network and many other excellent easy-to-use features offered within one converged network. Yet, it needs a lot of improvement in terms of giving 100% output.
For decades, the Voip industry has been working on addressing technical issues including Quality of service (QoS), interoperability, security, scalability, and integration with Public Switched Telephone Network (PSTN).
Voip systems need to be flexible enough to expand to cover large user market for both private and public services. Nevertheless, TCP/IP needs a lot of attention. To go back in history, the Internet protocols, including TCP, IP, and UDP were developed in the 70s and 80s era to support three major applications:
- file transfers using the File Transfer Protocol, or FTP
- electronic mail using the Simple Mail Transfer Protocol, or SMTP
- Remote host computer access using the TELNET protocol.
All these applications were data-oriented but not voice-orientated, and were therefore based upon IP’s connectionless network design. When TCP combined with IP, it enhanced the system’s reliability; however, it was not sufficient enough.
In order to support connection-oriented applications (TCP) over a connectionless environment (IP), additional protocols can enhance its functionality tenfold.
These include:
- Multicast Internet Protocol (Multicast IP) allows information transfer from a single source to multiple destinations. This helps in call conferencing.
- Real-time Transport Protocol (RTP) helps in functions such as payload identification, sequence numbering, and timestamps on the information.
- RTP Control Protocol (RTCP) monitors the quality and functionality of the RTP connection.
- Resource Reservation Protocol (RSVP) requests the network resources sharing, in order to provide ample bandwidth between sender and receiver.
- Real-Time Streaming Protocol (RTSP) supports the transfer and retrieval of real-time data from a media server or support for conferencing purposes.
- Session Description Protocol (SDP) communicates information about the media streams for a particular session, which includes session name, active session time, media (voice, video, etc.) to be used, the bandwidth required, and more.
- Session Announcement Protocol (SAP) packets sporadically carry out transmission to classify open sessions for the end user community.
Hence, proved that TCP/IP is still not adequate enough for VoIP, but once these new protocols are added, applications such as voice and video and the existing IP infrastructure can be enhanced greatly —supporting both connection-oriented and connectionless applications.