Research on VoIP: 10. Voice Protocol used in VoIP

Voice Protocol in VoIP

There are two types of protocol in VoIP:

Real-time Transport Protocol(RTP)

Provides the functinality to real time application
Runs on IP over UDP
has the capability of providing application framing by adding application-layers to datagrams.
It uses the sequence information to detemine whether packets are arrive in order.
It also provide time stamping to determine the packet interarrival time.
RTP uses typically informs a source about the quality of delivery. The source then adapt its sending rate accordingly.

Types of relays used by RTP

There are two types of relay used for transmission - mixer relay and translator.

Mixer relay

Combines data from two or more entries into single stream of data.
It can either retain or change the data format
Provides timming information for the combined stream of data and act as a source of timming synchronization.
Typically used for combining audio stream in real time application and service system that is not able to handle multiple stream.

Translator

A device that generate one or more RTP packrts for each incoming RTP packets.
Format of outing packets may be different from the incoimg packets.
Used in video application in which high quality siganl is converted to a lower quality in order to service receivers that support a lower data rate.
Some time, translator can be used to transfer a multicast packet to multiple destination and the RTP entries separated by an application-layer firewall.

RTP Packet Header

The following show the RTP header :

The header fields of the RTP packet are describe below:

Version - to indicate the version of the protocol
Padding - Indicate the existence of a padding field at the end of the payload. It is needed in application the need the payload to be multiple of some length.
Extension - indicate the use of an extension header for RTP
Constributing source count (CSC) - a 4 bit field that indicate the number of contributing source identifies.
Marker - boundaries in a stream of data traffic.
Paylaod - indicate the specfic type of RTP payload. Alos contain information about the use of compression or encryption.
Sequence - 16 bit field that indicates that a sender uses to identify a particular packet withihn a sequence of packets
Time stamp - to enable the receiver to rearrange back the original data.
Synchronization Source Identifier - randomly generated field to indicate the RTP source in an RTP session.
Contributing Source identifier - an optional field to show the contributing source for the data.

Real-time Transport Control Protocol

Runs on top of UDP.
Perform multicasting to provide feedback about data quality to all the members
The members can set an estimate of the perfromances of other members in the current active session.
Sender can send about data rates, quality of data transmission.
Receiver can send about packets loss rate, jitter variations and other problems encountered.
All session members must be able to evaluate the performances of other session members.

RTCP packet limitation

It msut be limited to less than 5%.
If it is more than that, it would affect the voice quality and the bandwidth of the channel.
The number of active members increased, the transmission rate of RTCP packets must be reduced

Format of RTCP packet

What are the reports send in RTCP session?

There are 3 reports- sender report , receiver report and sources description (SEDS)

A receiver and sender report consists of:

NTP timestamp - 64 bit field indicates to when a sender report was send.
RTP tiemstamp - 32 bit field used by receiver to sequence RTP packets
Sender packet count - 32 bit field represent number of RTP ocetets transmitted bt sender in the session.
Sender byte count - 32 bit feild represent number of RTP octets transmitted by the sender in the current session.

The SR packets includes 0 or more RR blocks. One receiver block is included for each sender from which the members has received during the session.

Research on VoIP