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Protocol Overview
This chapter overviews the VoIP transport protocol.
Voice over IP uses the Internet Protocol (IP) as an underlying transport base. Voice is digitized, converted to IP packets, and transmitted from point to point over an IP network.
Signaling
VoIP standards define numerous signaling protocols that are used to set up and carry out the calls, transmit information required to identify and locate remote callers, and negotiate carrier and endpoint capabilities. Different companies developed numerous different signaling protocols within the VoIP scope. SIP and H.323 are dominating the VoIP arena as the most commonly used signaling protocols.
Choosing one signaling protocol over another when developing a VoIP solution is a matter of a set service requirement and the choice of equipment. SIP is commonly chosen among the full_scale VoIP carriers to make use of the abundance of SIP_compatible VoIP devices including the numerous inexpensive SIP phones and adapters. When multimedia communication over IP networks is required, including video conferencing and data calls in addition to audio transmission, H.323 becomes the natural choice.
As defined by the scope of the signaling protocols, there are numerous potential problems that may arise because of compatibility or networking problems. The dreadful "unable to connect" problem lies frequently in the domain of signaling protocols. The two peers, once located, may be unable to connect because of compatibility problems between the two endpoints (various SIP implementations are especially prone to this problem) as well as the lack of the required features such as conference calling in one of the connecting devices.
A network analyzer should recognize and support both SIP and H.323 signaling protocols, allowing the detection of problems that occur on the signaling phase early during the implementation of a VoIP system. Throughout this white paper, we'll be illustrating the problems and solutions with the help of CommView and CommView for WiFi, software_based network analyzers for wired and wireless networks that include a VoIP analysis engine.
Streams
Once the peers are located and a connection is made, the streaming of voice packets begins to occur. In order for the voice conversations to sound natural, without echoes and delays, the voice packets must be transferred over the IP network in real_time. All VoIP standards use Real_time Transport Protocol (RTP) for streaming voice packets in real_time.
RTP standard does not define, and therefore does not require the use of any specific UDP port, leaving it up to endpoints to agree on a certain port to commence a voice call. The floating_port implementation makes it difficult to traverse firewalls, often requiring the use of dedicated STUN servers to synchronize the endpoints. A frequent VoIP connectivity problem occurs while attempting to send or receive voice packets via a more or less random port that is blocked by a firewall. A network analyzer allows the RTP streams associated with a specific signaling session to be clearly seen, as well as the IP addresses and the ports being used for the VoIP call, which helps faster and easier deployment of a VoIP system.
RTP does not use TCP protocol for transmitting voice packets. Despite the fact that TCP guarantees the delivery of the packets, its session initiation time and associated delays are unacceptable for transferring multimedia data in real_time. Therefore, UDP is the natural choice here. As there is no recovery for the packets not delivered to the recipient, a certain percentage of voice packets are typically lost. While VoIP does provide means to reconstruct the lost packets without significant loss of voice call quality, packet loss beyond a certain level starts to noticeably degrade the quality of the conversation. The illustration above displays the numbers of lost packets, allowing identifying network problems on the streaming phase.
RTP encapsulates additional information in every voice packet, including payload type identification to identify the type of content being transmitted, sequence numbering that is used to detect and identify the lost packets, and time stamping to allow synchronization and jitter calculations. The additional information is extremely handy when analyzing RTP streams in order to identify the source of quality issues.
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