RTP- Real Time Transport Protocol CSCE 5580 Computer Networks– Spring 2006 Presented by: Vandana Anand Archana Paka

Overview Purpose Services What it does not provide Architecture RTP data transport protocol State machine Event flows Message formats Issues Conclusions References Outline Vandana Archana

RTP - Overview RTP provides end-to-end network delivery services for the transmission of real-time media. Audio Video Conferencing IP telephony Games and real time monitoring RTP is defined in RFC 3550 (RFC 1889) Follows a principles outlined in a paper by Clark and Tennenhouse. Application Level Framing Integrated Layer Processing

RTP - Overview It can be used over various network technologies, IP/UDP, ATM/AAL5 and IPX. Often used over UDP. Two protocols make up RTP  RTP, a data transfer protocol (carries the data)  RTCP, control protocol (carries session. QoS Info)

RTP - Overview RTP can be used over both network services  Unicast Separate copies of the data are sent from the source to each destination.  Mulitcast The data is sent from the source only once and the network is responsible for transmitting the data to multiple locations.

RTP - Purpose The purpose of RTP is to facilitate,Delivery, Monitoring, Reconstruction, Mixing and Synchronization of data streams It provides support for real time conferencing of groups of any size within an Internet. It offers quality of service feedback from receivers to the multicast group. It determines the order of data to be presented. It synchronizes media stream from different sources.

RTP - Services Payload type identification  Identifies the type of data being sent. Time Stamping  Time stamping is used for the proper play out of the media and to reduce jitter. Sequencing – using sequence number  RTP can sequence those packets which arrive out of order at the receiving end.  The sequence number can also be used to identify the lost packets.

What it does not provide… RTP does not  Provide mechanism in time delivery. This requires the support of lower layers that have control over resources.  Provide Quality-of-Service  Guarantee delivery or prevent out-of-order delivery  It doesn’t not assume that the underlying network is reliable and delivers packets in sequence.

RTP – Architecture Application Level Framing  Recovery from lost data to be performed by the application. retransmission may not be appropriate. Lost data can be due to quality of delivery may be more useful for destination(s) to inform source about the quality of transmission  Application provide data for retransmission rather than the transport protocol. Sending application may re-compute lost data before sending The sending application can provide revised values rather than simply transmitting lost values.

RTP - Architecture Integrated Layer Processing  In a typical layered protocol architecture data units are sequentially processed by each layer  Integrated layer processing allows adjacent layers of the protocol stack to be tightly coupled  RTP is not complete by itself it requires application-layer and transport layer capabilities. RTP runs on top of UDP. UDP provides the port addressing functionality of the transport layer. RTP also contains the transport layer function like sequencing.

RTP – Data Transfer Protocol RTP data transfer protocol supports the transfer of real-time data among a number of participants in a session. Session is identified by  Participant IP address.  RTP port #, for media data.  RTCP port #, for control data.

RTP – Data Transfer Protocol Each RTP data unit must include  Source identifiers  Timestamp  Payload format It allows the use of two kind of relays  Mixers  Translators These are active application level devices residing on the network

Mixers A mixer is an RTP relay that receives streams of RTP packets from one or more source, combines these streams and forwards a new RTP packet stream to one or more destinations. Mixer can change the data format or simple perform the mixing function.

Translators Translator is a simple device that produces one or more outgoing RTP packet for each incoming RTP packet. It converts inputs and resends in a new format It replicates incoming packet and send it to a number of unicast destinations.

RTP- Applications Receiving Media Streams  Conferencing Applications : Need to be able to receive a media stream from an RTP session and render it on the console.  A telephone answering machine : Need to be able to receive a media stream and store it in a file.  Application that records a conversation or conference : Must be able to receive media stream from an RTP session and both render it and store it in a file. Transmitting Media Streams  RTP Server Applications: Need to t ransmit captured or stored media across the network.

Outline … continued Overview Purpose Services What it does not provide Architecture RTP data transport protocol State machine Event Flows Message Format Issues Conclusion

overview RTP does not have a standard TCP or UDP port that it communicates on. The only standard that it obeys is that UDP communications are done on an even port and the next higher odd port is used for RTP Control Protocol (RTCP) communications. Although there are no standards assigned, RTP is generally configured to use ports RTP only carries voice/video data. Call setup and tear-down is usually performed by the SIP protocol. The fact that RTP uses a dynamic port range makes it difficult for it to traverse firewalls. In order to get around this problem, it is often necessary to set up a STUN server.

A Typical Communication A normal RTP procedure between the client and the server is as follows: client->server: OPTIONS Server->client: OK client->server: DESCRIBE Server->client: OK client->server: SETUP Server->client: OK client->server: PLAY Server->client: OK client->server: PAUSE Server->client: OK client->server: PLAY Server->client: OK client->server: TEARDOWN Server->client: OK The Options and Describe command can be issued at any time. But SETUP must be issued before any PLAY, PAUSE and TEARDOWN command. After the TEARDOWN command, the client disconnects to the server.

Client Side State Machine Init: It represents the server state before receiving the SETUP command from client. Ready: This command triggers the state into ready state. Play: Here the server calls the RTP part of the program to encapsulate the data in the required file and send them via RTP to the client. Pause : The client can send PAUSE to the server after it begins playing the file.

Server Side State Machine

RTP – Packet Format

RTP - Header Video and audio payloads are sent separately Uses sequence number to synchronise audio and video once received IP Header UDP Header RTP Header RTP Video Payload IP Header UDP Header RTP Header RTP Audio Payload

RTP – Packet Details Version(2bits):Represents the Version of the RTP Padding(1 bit): Used to round off the bits. It means the packet contains one or more additional padding octets at the end which are not part of payload Extension(1 bit): If this bit is set, then the fixed header must be followed by the header extension CSRC count(4 bits): The number of CSRC (contributing sources) identifiers that follow the fixed header. Marker(1 bit): Allow significant events to be marked in the packet stream. Timestamp(32 bits): Reflects the sampling instant of the first octet in the RTP data packet

RTP – Packet Details Payload type(7 bits): Identifies the format of the RTP payload and determine its interpretations by applications Sequence Number(16 bits): Random number which is incremented by 1 for each RTP packet send. The packet receiver uses it to detect packet loss and to restore packet sequence. SSRC,Sync source(32 bits): A random ID to identify the sync sources. No two sync sources within an RTP session have the same SSRC identifier. CSRC(32 bits): The list of CSRC’s for the data contained in this packet.

Example showing where the message formats used…..

A videoconference in progress using a dedicated device RTP

Mixer and Translators in RTP Translator do not change SSRC or CSRC fields unlike mixers, go through the RTP header structure. RTP header structure

Performance Issues IN RTP RTP Scalability RTP Multiplexing

RTP Scalabilty Congestion. This is due to the small access bandwidths of the user as compared to the network bandwidth and particularly occurs when a large number of users join a multicast RTP session at roughly the same time. State Storage. In order to estimate the group size, hosts must listen to the multicast group and count the number of distinct end systems which sends packets,its unique identifier (SSRC) must be stored. Delay. As the group sizes grow, the time may become very large. This interval may easily exceed the duration of group membership. This means that timely reporting of QoS problems from a specific user will not occur, and the value of the actual reports is lost. BYE Floods. Just as users may simultaneously join a group, we can expect users may simultaneously leave a group. Premature Timeouts. The timeout value is set to five times the current transmission period. As group membership decreases (due to BYE's), the timeout value decreases with it. However, there is a delay between decreases in group membership and transmission of packets with the decreased interval. This delay makes it possible for a user to prematurely timeout because of a sudden drop in group membership.

RTP Multiplexing The principle mechanism by which new functionality is added to RTP is by the definition of payload formats. Payload formats specify an application specific sub-layer just above RTP. these formats provide additional functions needed for just that application. However, all of these formats describe how RTP is to be used for carrying a single media session between two endpoints. There are scenarios, however, where it is more desirable to multiplex many voice connections into the same transport level connection, and into the same packet. Consider the scenario where two standard telephone users wish to communicate, but the long distance portion of the call is to be carried over the Internet. This will require the use of VoIP Gateways which act as an application level bridge. Since gateways typically terminate large numbers of telephone calls, there is likely to be multiple voice calls between any pair of gateways. Currently, gateways place each voice call in a separate transport connection, and thus the data from each user goes in separate packets. Due to compression techniques, the payload sizes are typically quite small, and IP-related overhead is substantial, resulting in a large packet overhead. One proposed solution is to multiplex many voice calls into the same connection (assuming they are destined to the same remote gateway), and into the same packet. This drastically reduces overhead. Protocol structures for carrying multiple voice payloads in an RTP packet are being developed and the minimization of the overhead per user is being sought.

Security Issues in RTP Desired Security Features 1. Confidentiality 2. Integrity 3. Performance considerations of security features

Security Issues in RTP Security Features Provided by RTP 1. Confidentiality 2. Authenticity and Integrity 3. Key Management

Showing the Security Keys

Some Security Features

CONCLUSIONS RTP provides powerful instruments for adaptive Video Transmission. RTP can be considered user-space transport entities, but does not run as stand-alone process. Mixers and translators are stand-alone processes. They terminate TCP or UDP connections.  Can accomplish complex control features.  Complexity of the protocol/algorithm is not so bad, because there is little hard guarantee (It relies on TCP or application for hard guarantees).

References William Stallings, “Computer Networking with Internet Protocols and Technology”, Prentice Hall (2004). Colin Perkins, “RTP – Audio and Video For The Internet”, Addison Wesley (2003).

Questions What are the roles of the RTP timestamp and sequence numbers?  The timestamp is used to place the incoming audio and video packets in the correct timing order (playout delay compensation). The sequence number is mainly used to detect losses. Sequence numbers increase by one for each RTP packet transmitted, timestamps increase by the time "covered" by a packet.

Questions Is RTP a transport protocol or a kind of application protocol?  RTP has important properties of a transport protocol: it runs on end systems, it provides demultiplexing. It differs from transport protocols like TCP in that it (currently) does not offer any form of reliability or a protocol-defined flow/congestion control. However, it provides the necessary hooks for adding reliability, where appropriate, and flow/congestion control.

Questions Why does RTP rely on the applications to provide security measures?  Targeted communication is high-speed (Audio/Visual and Multicast) making recognition of an attack without specific application knowledge difficult.