Lecture 9: Multimedia Transmission Protocol

Hongli Luo, CEIT

Multimedia Transmission Protocol

RTSP RTP RTCP SIP Socket Programming

User Control of Streaming Media: RTSP HTTP does not target

multimedia content no commands for fast

forward, etc.RTSP: RFC 2326 Allow the media player

and server to exchange playback control information

Allows a media player to control the transmission of a media stream

client-server application layer protocol

user control: rewind, fast forward, pause, resume, repositioning, etc…

What it doesn’t do: doesn’t define how

audio/video is encapsulated for streaming over network

doesn’t restrict how streamed media is transported (UDP or TCP possible)

doesn’t specify how media player buffers audio/video

RTSP: out of band control

FTP uses an “out-of-band” control channel:

file transferred over one TCP connection.

control info (directory changes, file deletion, rename) sent over separate TCP connection

“out-of-band”, “in-band” channels use different port numbers

RTSP messages also sent out-of-band:

RTSP control messages use different port numbers than media stream: out-of-band. port 554 Over TCP or UDP

media stream is considered “in-band”.

Adopted by RealNetworks

FTP: the file transfer protocol

transfer file to/from remote host client/server model

client: side that initiates transfer (either to/from remote) server: remote host

ftp: RFC 959 ftp server: port 21

file transfer FTPserver

FTPuser

interface

FTPclient

local filesystem

remote filesystem

user at host

FTP: separate control, data connections

FTP client contacts FTP server at port 21, TCP is transport protocol

client authorized over control connection

client browses remote directory by sending commands over control connection.

when server receives file transfer command, server opens 2nd TCP connection (for file) to client

after transferring one file, server closes data connection.

server opens another TCP data connection to transfer another file.

FTPclient

FTPserver

TCP control connection

port 21

TCP data connectionport 20

The control session remains open throughout the duration of the user session

control connection: “out of band”

FTP server maintains “state”: current directory, earlier authentication

RTSP Example

Scenario: metafile communicated to web browser browser launches player player sets up an RTSP control connection, data

connection to streaming server

Metafile Example

<title>Twister</title> <session> <group language=en lipsync> <switch> <track type=audio e="PCMU/8000/1" src = "rtsp://audio.example.com/twister/audio.en/lofi"> <track type=audio e="DVI4/16000/2" pt="90 DVI4/8000/1" src="rtsp://audio.example.com/twister/audio.en/hifi"> </switch> <track type="video/jpeg" src="rtsp://video.example.com/twister/video"> </group> </session>

RTSP Operation

RTSP Exchange Example C: SETUP rtsp://audio.example.com/twister/audio RTSP/1.0 Transport: rtp/udp; compression; port=3056; mode=PLAY S: RTSP/1.0 200 1 OK Session 4231

C: PLAY rtsp://audio.example.com/twister/audio.en/lofi RTSP/1.0 Session: 4231 Range: npt=0- S: RTSP/1.0 200 2 OK Session 4231

C: PAUSE rtsp://audio.example.com/twister/audio.en/lofi RTSP/1.0 Session: 4231 Range: npt=37 S: RTSP/1.0 200 3 OK Session 4231

C: TEARDOWN rtsp://audio.example.com/twister/audio.en/lofi RTSP/1.0 Session: 4231 S: RTSP/1.0 200 4 OK Session 4231

Real-Time Protocol (RTP)

RTP specifies packet structure for packets carrying audio, video data Audio: PCM, GSM, MP3 Video: MPEG and h.263 Proprietary audio and

video formats RTP packet provides

payload type identification

packet sequence numbering

time stamping

RFC 3550 RTP runs in end systems RTP packets encapsulated

in UDP segments interoperability: if two

Internet phone applications run RTP, then they may be able to work together

RTP runs on top of UDP

RTP libraries provide transport-layer interface that extends UDP:

• port numbers, IP addresses• payload type identification• packet sequence numbering• time-stamping

RTP Example

consider sending 64 kbps PCM-encoded voice over RTP.

application collects encoded data in chunks, e.g., every 20 msec = 160 bytes in a chunk.

audio chunk + RTP header form RTP packet, which is encapsulated in UDP segment

RTP header indicates type of audio encoding in each packet sender can change

encoding during conference.

RTP header also contains sequence numbers, timestamps.

RTP and QoS

RTP does not provide any mechanism to ensure timely data delivery or other QoS guarantees.

RTP does not provide Timely delivery of data QoS guarantees Guarantee delivery of packets Prevention of out-of-order delivery of packets

RTP encapsulation is only seen at end systems (not) by intermediate routers. routers providing best-effort service, making

no special effort to ensure that RTP packets arrive at destination in timely matter.

RTP Header (12 bytes)

Payload Type (7 bits): Indicates type of encoding currently being used. If sender changes encoding in middle of conference, sender informs receiver via payload type field.

•Payload type 0: PCM mu-law, 64 kbps•Payload type 3, GSM, 13 kbps•Payload type 7, LPC, 2.4 kbps•Payload type 26, Motion JPEG•Payload type 31. H.261•Payload type 33, MPEG2 video

Sequence Number (16 bits): Increments by one for each RTP packet sent, and may be used to detect packet loss and to restore packet sequence.

RTP Header (2) Timestamp field (32 bytes long): sampling

instant of first byte in this RTP data packet Receiver can use it to remove packet jitter and to

provide synchronous playout for audio, timestamp clock typically increments by one

for each sampling period (for example, each 125 usecs for 8 KHz sampling clock)

if application generates chunks of 160 encoded samples, then timestamp increases by 160 for each RTP packet when source is active. Timestamp clock continues to increase at constant rate when source is inactive.

SSRC field (32 bits long): identifies source of RTP stream. Each stream in RTP session should have distinct SSRC.

Miscellaneous fields (9 bits)

Developing Software Applications with RTP

Two approaches to develop an RTP-based networked applications Incorporate RTP by hand –

• write the code that performs RTP encapsulation at the sender side and RTP decapsulation at the client side

Use existing RTP libraries (for C programmers) and Java classes (for Java programmers)

• The libraries and classes perform the encapsulation and decapsulation for the application

Incorporate RTP by hand - example

A server that encapsulates stored video frames into RTP packets grab video frame, add RTP headers to frame and generate an RTP

packet create UDP segments, send segments to UDP socket include seq numbers and time stamps The API is the standard UDP socket API

A client decapsulates the RTP packet and display the video frame

RTSP Client: issue setup/play/pause/teardown commands Server: accepts the requests and take actions

RTP does not mandate a specific port number. The application developer specifies the port

number for the two sides of the application.

Use existing Java RTP class to implement (or C RTP library for C programmers) to implement the RTP. The sender application provides

• media chunk, payload-type number, SSRC, timestamp, destination port number, destination IP

Java Media Framework (JMF) includes a complete RTP implementation

Real-Time Control Protocol (RTCP)

works in conjunction with RTP.

each participant in RTP session periodically transmits RTCP control packets to all other participants.

each RTCP packet contains sender and/or receiver reports report statistics useful

to application: # packets sent, # packets lost, interarrival jitter, etc.

feedback can be used to control performance sender may modify its

transmissions based on feedback

RTCP - Continued

each RTP session: typically a single multicast address; all RTP /RTCP packets belonging to session use multicast address.

RTP, RTCP packets distinguished from each other via distinct port numbers.

RTCP port number is set to be equal to the RTP port number plus one

to limit traffic, each participant reduces RTCP traffic as number of conference participants increases

RTCP Packets

Receiver report packets: Receiver aggregates its reception report into a

single RTCP packet The packet is sent into the multicast tree that

connects all the session’s participants. Fields in reception report:

SSRC of RTP stream fraction of packets lost – the sender can switch to

different encoding rates last sequence number average interarrival jitter – a smoothed estimate of the

variation in the interarrival time between successive packets in the RTP stream

RTCP Packets

Sender report packets: Sender creates and transmits RTCP sender

report packets The packets include information such as

SSRC of RTP stream, Time stamp, wall clock time (current time) of the most

recently generated RTP packet in the stream number of packets sent, number of bytes sent

Sender reports can be used to synchronize different media streams within a RTP session.

RTCP Packets

Source description packets: Sender also creates and transmits source

description packets. Includes e-mail address of sender, sender's

name, SSRC of associated RTP stream, application that generates the RTP stream

provide mapping between the SSRC and the user/host name

RTCP packets are stackable Receiver reception reports, sender reports, and source

descriptors can be concatenated into a single packet The RTCP packet is then encapsulated into a UDP

segment

Synchronization of Streams

RTCP can synchronize different media streams within a RTP session

consider videoconferencing app for which each sender generates one RTP stream for video, one for audio.

timestamps in RTP packets tied to the video, audio sampling clocks not tied to wall-clock

time (real time)

each RTCP sender-report packet contains (for most recently generated packet in associated RTP stream): timestamp of RTP packet wall-clock time for when

packet was created. receivers uses

association to synchronize playout of audio, video

RTCP Bandwidth Scaling

RTCP attempts to limit its traffic to 5% of session bandwidth.

Example Suppose one sender,

sending video at 2 Mbps. Then RTCP attempts to limit its traffic to 100 Kbps.

RTCP gives 75% of rate to receivers; remaining 25% to sender

75 kbps is equally shared among receivers: with R receivers, each

receiver gets to send RTCP traffic at 75/R kbps.

sender gets to send RTCP traffic at 25 kbps.

participant determines RTCP packet transmission period by calculating avg RTCP packet size (across entire session) and dividing by allocated rate

RTCP Bandwidth Scaling (2)

The period for transmitting RTCP packets for a sender is

T = (number of senders ) * (avg. RTCP packet size)

/ (.25 * .05 * session bandwidth)

The period for transmitting RTCP packets for a receiver is

T = (number of senders ) *(avg. RTCP packet size)/ (.75 * .05 * session bandwidth)

SIP: Session Initiation Protocol [RFC 3261]

SIP long-term vision:

all telephone calls, video conference calls take place over Internet

people are identified by names or e-mail addresses, rather than by phone numbers

you can reach callee, no matter where callee roams, no matter what IP device callee is currently using – computer or PDA

SIP Services

Setting up a call between caller and callee, SIP provides mechanisms .. for caller to let

callee know she wants to establish a call

so caller, callee can agree on media type, encoding

to end call

determine current IP address of callee: Callee has dynamic IP

by DHCP or has multiple IP devices

maps mnemonic identifier to current IP address

call management: add new media

streams during call change encoding

during call invite others transfer, hold calls

Setting up a call to known IP address Alice’s SIP invite message indicates her port number, IP address, encoding she prefers to receive (PCM ulaw)

Bob’s 200 OK message indicates his port number, IP address, preferred encoding (GSM)

SIP messages can be sent over TCP or UDP; here sent over RTP/UDP. default SIP port number is 5060.

time time

Bob'stermina l rings

A lice

167.180.112.24

Bob

193.64.210.89

port 38060

Law audio

G SMport 48753

Setting up a call (more) SIP is an out-of-band

protocol SIP messages are sent and

received in sockets different from those for media data

SIP messages are ASCII-readable and resemble HTTP messages

SIP requires all messages to be acknowledged It can run over UDP or TCP

media can be sent over RTP or some other protocol

codec negotiation: suppose Bob doesn’t

have PCM ulaw encoder.

Bob will instead reply with 606 Not Acceptable Reply, listing his encoders Alice can then send new INVITE message, advertising different encoder

rejecting a call Bob can reject with

replies “busy,” “gone,” “payment required,” “forbidden”

SIP Addresses

Bob’s SIP address is sip:bob@193.64.210.89 When Alice’s SIP device sends an INVITE

message, the message would include this email-like address

The SIP infrastructure would then route the message to the IP advice that Bob is currently using

Other possible forms for SIP address Phone number First/last name

SIP address can be included in Web page

Example of SIP message

INVITE sip:bob@domain.com SIP/2.0

Via: SIP/2.0/UDP 167.180.112.24

From: sip:alice@hereway.com

To: sip:bob@domain.com

Call-ID: a2e3a@pigeon.hereway.com

Content-Type: application/sdp

Content-Length: 885

c=IN IP4 167.180.112.24

m=audio 38060 RTP/AVP 0

Notes: HTTP message syntax sdp = session description protocol Call-ID is unique for every call.

Here we don’t know Bob’s IP address. Intermediate SIPservers needed.

Alice sends, receives SIP messages using SIP default port 506

Alice specifies in Via: IP address of the device, header that SIP client sends, receives SIP messages over UDP

Name translation and user locataion

caller wants to call callee, but only has callee’s name or e-mail address.

need to get IP address of callee’s current host: user moves around DHCP protocol user has different IP

devices (PC, PDA, car device)

result can be based on: time of day (work,

home) caller (don’t want boss to

call you at home) status of callee (calls

sent to voicemail when callee is already talking to someone)

Service provided by SIP servers:

SIP proxy server SIP registrar server

SIP Proxy

Alice sends invite message to her proxy server contains address sip:bob@domain.com

proxy responsible for routing SIP messages to callee possibly through multiple proxies.

callee sends response back through the same set of proxies.

proxy returns SIP response message to Alice contains Bob’s IP address

proxy analogous to local DNS server

SIP Registrar

REGISTER sip:domain.com SIP/2.0

Via: SIP/2.0/UDP 193.64.210.89

From: sip:bob@domain.com

To: sip:bob@domain.com

Expires: 3600

when Bob starts SIP client, client sends SIP REGISTER message to Bob’s registrar server

(similar function needed by Instant Messaging) Often SIP registrars and SIP proxies are run on

the same host

Register Message:

ExampleCaller jim@umass.edu with places a call to keith@upenn.edu

(1) Jim sends INVITEmessage to umass SIPproxy. (2) Proxy forwardsrequest to upenn registrar server. (3) upenn server returnsredirect response,indicating that it should try keith@eurecom.fr

(4) umass proxy sends INVITE to eurecom registrar. (5) eurecom registrar forwards INVITE to 197.87.54.21, which is running keith’s SIP client. (6-8) SIP response sent back (9) media sent directly between clients. Note: also a SIP ack message, which is not shown.

SIP client217.123.56.89

SIP client197.87.54.21

SIP proxyum ass.edu

SIP registrarupenn.edu

SIPregistrareurecom .fr

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5

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Comparison with H.323

H.323 is another signaling protocol for real-time, interactive audio and video conferencing

H.323 is a complete, vertically integrated suite of protocols for multimedia conferencing: signaling, registration, admission control, transport, codecs

SIP is a single component. Works with RTP, but does not mandate it. Can be combined with other protocols, services

H.323 comes from the ITU (telephony).

SIP comes from IETF: Borrows much of its concepts from HTTP SIP has Web flavor,

whereas H.323 has telephony flavor.

SIP uses the KISS principle: Keep it simple stupid.

Socket programming

Development of network applications Implementation of protocol standard defined in an RFC

• Client and server conform to the rules of RFC• Use the port number associated with the protocol• Allows interoperability

Proprietary network application• The application-layer protocol used by the client

and the server do not necessarily conform to any existing RFC

• Developer creates both client and server programs• Not interoperable with other applications• Not to use well-known port numbers defined in

RFCs• TCP or UDP at the transport layer?

Socket programming

Socket API introduced in BSD4.1 UNIX,

1981 explicitly created, used,

released by apps client/server paradigm two types of transport service

via socket API: unreliable datagram reliable, byte stream-

oriented

a host-local, application-created,

OS-controlled interface (a “door”) into which

application process can both send and

receive messages to/from another

application process

socket

Goal: learn how to build client/server application that communicate using sockets

Socket-programming using TCP

Socket: a door between application process and end-end-transport protocol (UCP or TCP)

TCP service: reliable transfer of bytes from one process to another

process

TCP withbuffers,

variables

socket

controlled byapplicationdeveloper

controlled byoperating

system

host orserver

process

TCP withbuffers,

variables

socket

controlled byapplicationdeveloper

controlled byoperatingsystem

host orserver

internet

Socket programming with TCP

Client must contact server server process must first be

running server must have created

socket (door) that welcomes client’s contact

Client contacts server by: creating client-local TCP

socket specifying IP address, port

number of server process The client choses a source

port number When client creates socket:

client TCP establishes connection to server TCP

When contacted by client, server TCP creates new socket for server process to communicate with client

allows server to talk with multiple clients

source port numbers used to distinguish clients

TCP socket is identified by a four-tuple: (source IP address, source port number, destination IP address, destination port number)

TCP provides reliable, in-order transfer of bytes (“pipe”) between client and server

application viewpoint

Client/server socket interaction: TCP

wait for incomingconnection requestconnectionSocket =welcomeSocket.accept()

create socket,port=x, forincoming request:welcomeSocket =

ServerSocket()

create socket,connect to hostid, port=xclientSocket =

Socket()

closeconnectionSocket

read reply fromclientSocket

closeclientSocket

Server (running on hostid) Client

send request usingclientSocketread request from

connectionSocket

write reply toconnectionSocket

TCP connection setup

outT

oSer

ver

to network from network

inFr

omS

erve

r

inFr

omU

ser

keyboard monitor

Process

clientSocket

inputstream

inputstream

outputstream

TCPsocket

Clientprocess

client TCP socket

Stream jargon A stream is a sequence of

characters that flow into or out of a process.

An input stream is attached to some input source for the process, e.g., keyboard or socket.

An output stream is attached to an output source, e.g., monitor or socket.

Socket programming with TCP

Example client-server app:1) client reads line from standard

input (inFromUser stream) , sends to server via socket (outToServer stream)

2) server reads line from socket3) server converts line to

uppercase, sends back to client

4) client reads, prints modified line from socket (inFromServer stream)

Example: Java client (TCP)

import java.io.*; import java.net.*; class TCPClient {

public static void main(String argv[]) throws Exception { String sentence; String modifiedSentence;

BufferedReader inFromUser = new BufferedReader(new InputStreamReader(System.in));

Socket clientSocket = new Socket("hostname", 6789); System.out.println(“client port: " + clientSocket.getLocalPort()); DataOutputStream outToServer = new DataOutputStream(clientSocket.getOutputStream());

Createinput stream

Create client socket,

connect to server

Createoutput stream

attached to socket

Example: Java client (TCP), cont.

BufferedReader inFromServer = new BufferedReader(new InputStreamReader(clientSocket.getInputStream()));

sentence = inFromUser.readLine();

outToServer.writeBytes(sentence + '\n');

modifiedSentence = inFromServer.readLine();

System.out.println("FROM SERVER: " + modifiedSentence);

clientSocket.close(); } }

Createinput stream

attached to socket

Send lineto server

Read linefrom server

Example: Java server (TCP)

import java.io.*; import java.net.*;

class TCPServer {

public static void main(String argv[]) throws Exception { String clientSentence; String capitalizedSentence;

ServerSocket welcomeSocket = new ServerSocket(6789); while(true) { Socket connectionSocket = welcomeSocket.accept();

BufferedReader inFromClient = new BufferedReader(new InputStreamReader(connectionSocket.getInputStream()));

Createwelcoming socket

at port 6789

Wait, on welcomingsocket for contact

by client

Create inputstream, attached

to socket

Example: Java server (TCP), cont

DataOutputStream outToClient = new DataOutputStream(connectionSocket.getOutputStream());

clientSentence = inFromClient.readLine();

capitalizedSentence = clientSentence.toUpperCase() + '\n';

outToClient.writeBytes(capitalizedSentence); } } }

Read in linefrom socket

Create outputstream,

attached to socket

Write out lineto socket

End of while loop,loop back and wait foranother client connection

Socket programming with UDP

UDP: no “connection” between client and server

no handshaking sender explicitly attaches IP

address and port of destination to each packet

server must extract IP address, port of sender from received packet

UDP: transmitted data may be received out of order, or lost

application viewpoint

UDP provides unreliable transfer of groups of bytes (“datagrams”)

between client and server

Client/server socket interaction: UDP

closeclientSocket

Server (running on hostid)

read reply fromclientSocket

create socket,clientSocket = DatagramSocket()

Client

Create, address (hostid, port=x,send datagram request using clientSocket

create socket,port=x, forincoming request:serverSocket = DatagramSocket()

read request fromserverSocket

write reply toserverSocketspecifying clienthost address,port number

Example: Java client (UDP)

sendP

ack

et

to network from network

rece

iveP

ack

et

inF

rom

Use

r

keyboard monitor

Process

clientSocket

UDPpacket

inputstream

UDPpacket

UDPsocket

Output: sends packet (recallthat TCP sent “byte stream”)

Input: receives packet (recall thatTCP received “byte stream”)

Clientprocess

client UDP socket

Example: Java client (UDP)

import java.io.*; import java.net.*; class UDPClient { public static void main(String args[]) throws Exception { BufferedReader inFromUser = new BufferedReader(new InputStreamReader(System.in)); DatagramSocket clientSocket = new DatagramSocket(); InetAddress IPAddress = InetAddress.getByName("hostname"); byte[] sendData = new byte[1024]; byte[] receiveData = new byte[1024]; String sentence = inFromUser.readLine();

sendData = sentence.getBytes();

Createinput stream

Create client socket

Translate hostname to IP

address using DNS

Example: Java client (UDP), cont.

DatagramPacket sendPacket = new DatagramPacket(sendData, sendData.length, IPAddress, 9876); clientSocket.send(sendPacket); DatagramPacket receivePacket = new DatagramPacket(receiveData, receiveData.length); clientSocket.receive(receivePacket); String modifiedSentence = new String(receivePacket.getData()); System.out.println("FROM SERVER:" + modifiedSentence); clientSocket.close(); }

}

Create datagram with data-to-send,

length, IP addr, port

Send datagramto server

Read datagramfrom server

Example: Java server (UDP)

import java.io.*; import java.net.*; class UDPServer { public static void main(String args[]) throws Exception { DatagramSocket serverSocket = new DatagramSocket(9876); byte[] receiveData = new byte[1024]; byte[] sendData = new byte[1024]; while(true) { DatagramPacket receivePacket = new DatagramPacket(receiveData, receiveData.length);

serverSocket.receive(receivePacket);

Createdatagram socket

at port 9876

Create space forreceived datagram

Receivedatagra

m

Example: Java server (UDP), cont

String sentence = new String(receivePacket.getData()); InetAddress IPAddress = receivePacket.getAddress(); int port = receivePacket.getPort(); String capitalizedSentence = sentence.toUpperCase();

sendData = capitalizedSentence.getBytes(); DatagramPacket sendPacket = new DatagramPacket(sendData, sendData.length, IPAddress, port); serverSocket.send(sendPacket); } }

}

Get IP addrport #, of

sender

Write out datagramto socket

End of while loop,loop back and wait foranother datagram

Create datagramto send to client