[Network]Multimedia Networking

Principle

1 Multimedia Networking Applications

1.1 Classifications

1) stored streaming

2) live streaming

3) interactive real-time

1.2 Characteristics

1) delay 

2) loss

1.3 Streaming

client playout begins all the data has arrived.

1.4 Stored Streaming

client can pause, rewind, and push the slider bar.

1.5 Live Streaming

Much same as stored streaming, but forward fast!

1.6 Real-Time

end-end delay requirement

1.7 QoS 

Use application-level techniques to mitigate effect of delay and loss.

2 Multimedia need

2.1 Client-Side Buffering

As we can see, the streaming from server to client go through: transmission -> network delay -> client reception and buffering(playout delay) -> playout at client

I think, the buffer of client is like a list. The streaming fills in the list and client play out the files.

2.2 Use of UDP versus TCP

1. Client rate

different client receive rate -> server stores, transmits multiple copies of video encoded at different rates.

2. Use UDP to avoid TCP congestion control for time-sensitive traffic

2.3 Encoding: RTSP - User Control of Streaming Media

C/S Application Layer Protocol

User Control: pause, rewind, fast forward, repositioning, etc.

Example: the client request by HTTP GET method to server. Below I show some request types of RTSP Operations

SETUP: setup the connection

PLAY: play command

PAUSE: pause command

3 Best out of Best Effort Service

3.1 Playout delay

1. Goal: Minimize playout delay, keeping late loss rate low

2. Approach

Estimate network delay, adjust playout delay at the beginning of each talk spurt.

How determine whether a packet is first in a talkspurt

1) successive time stamp > 20 ms

2) sequence num without gaps

Talk spurt begins.

3.2 Recovery from Packet Loss

1. Forward Error Correction(FEC)

1) idea: Except retransmit the error packet, the FEC can correct the error data and recover it.

3.3 Content Delivery Network (CDN)

1. Idea

Contents will exist on several servers.

4. Tricks of MM

1. Use UDP

2. client-side: adaptive playout delay

3. server-side: match bandwidth 

4. FEC: recovery

5. CDN: bring content close to client

Protocols: Application-Level

1 Real-Time Protocol-RTP-Top of UDP in Transport Layer Protocol

Get the multimedia data encapsulated from Application layer, and it add some information for the data in its head.

1.1 RTP Header

1. Payload Type: data type - GSM, JPEG...

2. Sequence Number: detect loss, and restore packet sequence

3. Timestamp

4. SSRC: each stream in RTP session should have distinct SSRC

1.2 Programming

1) grab video frame

2) include RTP Seq num and time stamp

3) form head and send segments to UDP socket.

2 Real-Time Control Protocol - RTCP

Add reports from sender and receiver as feedback to control performance, such as playout delay adjustment and bandwidth adjustment.

Report: 

sender: time, seq num

receiver: last packet num, average delay...

Synchronization: in report packet

time stamp

wall-clock time

3 Session Initiation Protocol - SIP

Setting up a call and agree on media type and encoding(can build new encoder).

Can be sent on UDP or TCP

Service provided by SIP servers:

1. SIP registrar server

2. SIP Proxy server

1-8: register

9: communication

4 Providing Multiple Classes of Service

Principles:

1. Principle 1: router need packet marked to distinguish different classes: mixture need

2. Principle 2: provide protection for one class from others: bandwidth allocation

3. Principle 3: use resource as efficiently as possible: bandwidth is not used.

Scheduling:

1. FIFO: 

discard policy: tail, priority, random

2. Priority:

3. Round Robin Scheduling: class round

Policing Mechanism: 

limit traffic not to exceed some parameters

One minute sent packets

1. Average Rate

2. Peak Rate

3. Burst Size:

Implementation: token bucket

5 QoS Guarantees

1. block call if it cannot meet needs

2. resource reservation