Free Essay

Multimedia

In: Computers and Technology

Submitted By ameya91
Words 4574
Pages 19
MPSTME, NMIMS

2012-2013
Multimedia
Ameya Dighe 162, Raghav Jaju 165, Ujjwal Kumar 177

Times have changed. People want to use the Internet not only for text and image communications, but also for audio and video services. We concentrate on applications that use the Internet for audio and video services.
1. INTRODUCTION
Audio & video services is divided in 3 parts: * streaming stored audio/video, * streaming live audio/video, * interactive audio/video

In the first category, streaming stored audio/video, the files are compressed and stored on a server. A client downloads the files through the Internet. This is sometimes referred to as on-demand audio/video. Examples of stored audio files are songs, symphonies, books on tape, and famous lectures. Examples of stored video files are movies, TV shows, and music video clips.

In the second category, streaming live audio/video, a user listens to broadcast audio and video through the Internet. A good example of this type of application is the Internet radio. Some radio stations broadcast their programs only on the Internet; many broadcast them both on the Internet and on the air. Internet TV is not popular yet, but many people believe that TV stations will broadcast their programs on the Internet in the future.

In the third category, interactive audio/video, people use the Internet to interactively communicate with one another. A good example of this application is Internet telephony and Internet teleconferencing.Example includes Skype or facebook video chats.
2. DIGITIZING AUDIO AND VIDEO
DIGITIZING AUDIO
When sound is fed into a microphone, an electronic analog signal is generated that represents the sound amplitude as a function of time. The signal is called an analog audiosignal. An analog signal, such as audio, can be digitized to produce a digital signal. According to the Nyquist theorem, if the highest frequency of the signal is f, we need to sample the signal 2f times per second. There are other methods for digitizing an audio signal, but the principle is the same.

Sampling | Voice | Music | Samples per second | 8000 | 44100 | Bits per sample | 8 | 16 | Result | 64kbps | 705.6 kbps for monaural1.411 MBPS for stereo |

Digitizing Video
A video consists of a sequence of frames. If the frames are displayed on the screen fast enough, we get an impression of motion. The reason is that our eyes cannot distinguish the rapidly flashing frames as individual ones. There is no standard number of frames per second; in North America 25 frames per second is common. However, to avoid a condition known as flickering, a frame needs to be refreshed. The TV industry repaints each frame twice. This means 50 frames need to be sent, or if there is memory at the sender site, 25 frames with each frame repainted from the memory.
Each frame is divided into small grids, called picture elements or pixels. For blackand-white TV, each 8-bit pixel represents one of 256 different gray levels. For a color TV, each pixel is 24 bits, with 8 bits for each primary color (red, green, and blue). We can calculate the number of bits in a second for a specific resolution. In the lowest resolution a color frame is made of 1,024768 pixels. This means that we need

2*25*1024*768*24=944 MBPS This data rate needs a very high data rate technology such as SONET. To send video using lower-rate technologies, we need to compress the video.

Compression is needed to send video over the Internet.
3. STREAMING STORED AUDIO/VIDEO
Now that we have discussed digitizing and compressing audio/video, we turn our attention to specific applications. The first is streaming stored audio and video. Downloading these types of files from a Web server can be different from downloading other types of files. To understand the concept, let us discuss three approaches, each with a different complexity.

First Approach: Using a Web Server

A compressed audio/video file can be downloaded as a text file. The client (browser) can use the services of HTTP and send a GET message to download the file. The Web server can send the compressed file to the browser. The browser can then use a help application, normally called a media player, to play the file. Figure below shows this approach. This approach is very simple and does not involve streaming. However, it has a drawback. An audio/video file is usually large even after compression. An audio file may contain tens of megabits, and a video file may contain hundreds of megabits. In this approach, the file needs to download completely before it can be played. Using contemporary data rates, the user needs some seconds or tens of seconds before the file can be played.

Second Approach: Using a Web Server with Metafile
In another approach, the media player is directly connected to the Web server for downloading the audio/video file. The Web server stores two files: the actual audio/video file and a metafile that holds information about the audio/video file. Figure below shows the steps in this approach.

1. The HTTP client accesses the Web server using the GET message.
2. The information about the metafile comes in the response.
3. The metafile is passed to the media player.
4. The media player uses the URL in the metafile to access the audio/video file.
5. The Web server responds.

Third Approach: Using a Media Server
The problem with the second approach is that the browser and the media player both use the services of HTTP. HTTP is designed to run over TCP. This is appropriate for retrieving the metafile, but not for retrieving the audio/video file. The reason is that TCP retransmits a lost or damaged segment, which is counter to the philosophy of streaming. We need to dismiss TCP and its error control; we need to use UDP. However, HTTP, which accesses the Web server, and the Web server itself are designed for TCP; we need another server, a media server. Figure 25 below shows the concept.

1. The HTTP client accesses the Web server using a GET message.
2. The information about the metafile comes in the response.
3. The metafile is passed to the media player.
4. The media player uses the URL in the metafile to access the media server to download the file. Downloading can take place by any protocol that uses UDP.
5. The media server responds.

Fourth Approach: Using a Media Server and RTSP
The Real-Time Streaming Protocol (RTSP) is a control protocol designed to add more functionalities to the streaming process. Using RTSP, we can control the playing of audio/video. RTSP is an out-of-band control protocol that is similar to the second connection in FTP. Figure below shows a media server and RTSP.

1. The HTTP client accesses the Web server using a GET message.
2. The information about the metafile comes in the response.
3. The metafile is passed to the media player. 4. The media player sends a SETUP message to create a connection with the media server.
5. The media server responds.
6. The media player sends a PLAY message to start playing (downloading).
7. The audio/video file is downloaded using another protocol that runs over UDP.
8. The connection is broken using the TEARDOWN message.
9. The media server responds.
The media player can send other types of messages. For example, a PAUSE message temporarily stops the downloading; downloading can be resumed with a PLAY message.

STREAMING LIVE AUDIO/VIDEO

* Streaming live audio/video is similar to the broadcasting of audio and video by radio and TV stations. * Instead of broadcasting to the air, the stations broadcast through the Internet. There are several similarities between streaming stored audio/video and streaming live audio/video. * However, there is a difference. In the first application, the communication is unicast and on-demand. In the second, the communication is multicast and live. * However, presently, live streaming is still using TCP and multiple unicasting instead of multicasting. There is still much progress to be made in this area.

REAL-TIME INTERACTIVE AUDIO/VIDEO
In real-time interactive audio/video, people communicate with one another in real time. The Internet phone or voice over IP is an example of this type of application. Video conferencing is another example that allows people to communicate visually and orally.

Characteristics
Before discussing the protocols used in this class of applications, we discuss some characteristics of real-time audio/video communication.

Time Relationship

* Real-time data on a packet-switched network require the preservation of the time relationship between packets of a session. * For example, let us assume that a real-time video server creates live video images and sends them online. The video is digitized and packetized. There are only three packets, and each packet holds 10 s of video information.

Timestamp

* One solution to jitter is the use of a timestamp. If each packet has a timestamp that shows the time it was produced relative to the first (or previous) packet, then the receiver can add this time to the time at which it starts the playback..

Playback Buffer

* To be able to separate the arrival time from the playback time, we need a buffer to store the data until they are played back. The buffer is referred to as a playback buffer. When a session begins (the first bit of the first packet arrives), the receiver delays playing the data until a threshold is reached.

* Data are stored in the buffer at a possibly variable rate, but they are extracted and played back at a fixed rate. Note that the amount of data in the buffer shrinks or expands, but as long as the delay is less than the time to play back the threshold amount of data, there is no jitter.

Ordering

* In addition to time relationship information and timestamps for real-time traffic, one more feature is needed. We need a sequence number for each packet. * The timestamp alone cannot inform the receiver if a packet is lost. For example, suppose the timestamps are 0, 10, and 20. If the second packet is lost, the receiver receives just two packets with timestamps 0 and 20. The receiver assumes that the packet with timestamp 20 is the second packet, produced 20 s after the first. * The receiver has no way of knowing that the second packet has actually been lost. A sequence number to order the packets is needed to handle this situation.

Multicasting

* Multimedia play a primary role in audio and video conferencing. The traffic can be heavy, and the data are distributed using multicasting methods. Conferencing requires two-way communication between receivers and senders.

Translation

* Sometimes real-time traffic needs translation. A translator is a computer that can change the format of a high-bandwidth video signal to a lower-quality narrow bandwidth signal. * This is needed, for example, for a source creating a high-quality video signal at 5 Mbps and sending to a recipient having a bandwidth of less than 1 Mbps. To receive the signal, a translator is needed to decode the signal and encode it again at a lower quality that needs less bandwidth.
Mixing

* If there is more than one source that can send data at the same time (as in a video or audio conference), the traffic is made of multiple streams. * To converge the traffic to one stream, data from different sources can be mixed. A mixer mathematically adds signals coming from different sources to create one single signal.

4. STREAMING LIVE AUDIO/VIDEO

Streaming live audio/video is similar to the broadcasting of audio and video by radio and TV stations. Instead of broadcasting to the air, the stations broadcast through the Internet. There are several similarities between streaming stored audio/video and streaming live audio/video. They are both sensitive to delay; neither can accept retransmission. However, there is a difference. In the first application, the communication is unicast and on-demand. In the second, the communication is multicast and live. Live streaming is better suited to the multicast services of IP and the use of protocols such as UDP and RTP (discussed later). However, presently, live streaming is still using TCP and multiple unicasting instead of multicasting. There is still much progress to be made in this area.

5. REAL-TIME INTERACTIVE AUDIO/VIDEO
In real-time interactive audio/video, people communicate with one another in real time. The Internet phone or voice over IP is an example of this type of application. Video conferencing is another example that allows people to communicate visually and orally.

Characteristics
Before discussing the protocols used in this class of applications, we discuss some characteristics of real-time audio/video communication.

Time Relationship
Real-time data on a packet-switched network require the preservation of the time relationship between packets of a session. For example, let us assume that a real-time video server creates live video images and sends them online. The video is digitized and packetized. There are only three packets, and each packet holds 10 s of video information.
The first packet starts at 00:00:00, the second packet starts at 00:00:10, and the third packet starts at 00:00:20. Also imagine that it takes 1 s (an exaggeration for simplicity) for each packet to reach the destination (equal delay). The receiver can play back the first packet at 00:00:01, the second packet at 00:00:11, and the third packet at 00:00:21.
Although there is a 1-s time difference between what the server sends and what the client sees on the computer screen, the action is happening in real time. The time relationship between the packets is preserved. The 1-s delay is not important. Figure below shows the idea.

But what happens if the packets arrive with different delays? For example, the first packet arrives at 00:00:01 (1-s delay), the second arrives at 00:00:15 (5-s delay), and the third arrives at 00:00:27 (7-s delay). If the receiver starts playing the first packet at 00:00:01, it will finish at 00:00:11. However, the next packet has not yet arrived; it arrives 4 s later. There is a gap between the first and second packets and between the second and the third as the video is viewed at the remote site. This phenomenon is called jitter. Figure below shows the situation.

Timestamp

One solution to jitter is the use of a timestamp. If each packet has a timestamp that shows the time it was produced relative to the first (or previous) packet, then the receiver can add this time to the time at which it starts the playback. In other words, the receiver knows when each packet is to be played. Imagine the first packet in the previous example has a timestamp of 0, the second has a timestamp of 10, and the third a timestamp of 20. If the receiver starts playing back the first packet at 00:00:08, the second will be played at 00:00:18, and the third at 00:00:28. There are no gaps between the packets. Figure below shows the situation.

Playback Buffer

To be able to separate the arrival time from the playback time, we need a buffer to store the data until they are played back. The buffer is referred to as a playback buffer. When a session begins (the first bit of the first packet arrives), the receiver delays playing the data until a threshold is reached. In the previous example, the first bit of the first packet arrives at 00:00:01; the threshold is 7 s, and the playback time is 00:00:08. The threshold is measured in time units of data. The replay does not start until the time units of data are equal to the threshold value.

Data are stored in the buffer at a possibly variable rate, but they are extracted and played back at a fixed rate. Note that the amount of data in the buffer shrinks or expands, but as long as the delay is less than the time to play back the threshold amount of data, there is no jitter. Figure below shows the buffer at different times for our example.

Ordering
In addition to time relationship information and timestamps for real-time traffic, one more feature is needed. We need a sequence number for each packet. The timestamp alone cannot inform the receiver if a packet is lost. For example, suppose the timestamps are 0, 10, and 20. If the second packet is lost, the receiver receives just two packets with timestamps 0 and 20. The receiver assumes that the packet with timestamp 20 is the second packet, produced 20 s after the first. The receiver has no way of knowing that the second packet has actually been lost. A sequence number to order the packets is needed to handle this situation.

Multicasting
Multimedia play a primary role in audio and video conferencing. The traffic can be heavy, and the data are distributed using multicasting methods. Conferencing requires two-way communication between receivers and senders.

Translation
Sometimes real-time traffic needs translation. A translator is a computer that can change the format of a high-bandwidth video signal to a lower-quality narrow bandwidth signal. This is needed, for example, for a source creating a high-quality video signal at 5 Mbps and sending to a recipient having a bandwidth of less than 1 Mbps. To receive the signal, a translator is needed to decode the signal and encode it again at a lower quality that needs less bandwidth.

Mixing
If there is more than one source that can send data at the same time (as in a video or audio conference), the traffic is made of multiple streams. To converge the traffic to one stream, data from different sources can be mixed. A mixer mathematically adds signals coming from different sources to create one single signal.

Support from Transport Layer Protocol
The procedures mentioned in the previous sections can be implemented in the application layer. However, they are so common in real-time applications that implementation in the transport layer protocol is preferable. Let’s see which of the existing transport layers is suitable for this type of traffic. TCP is not suitable for interactive traffic. It has no provision for timestamping, and it does not support multicasting. However, it does provide ordering (sequence numbers). One feature of TCP that makes it particularly unsuitable for interactive traffic is its error control mechanism. In interactive traffic, we cannot allow the retransmission of a lost or corrupted packet. If a packet is lost or corrupted in interactive traffic, it must just be ignored. Retransmission upsets the whole idea of timestamping and playback. Today there is so much redundancy in audio and video signals (even with compression) that we can simply ignore a lost packet. The listener or viewer at the remote site may not even notice it.

UDP is more suitable for interactive multimedia traffic. UDP supports multicasting and has no retransmission strategy. However, UDP has no provision for timestamping, sequencing, or mixing. A new transport protocol, Real Time Transport Protocol (RTP), provides these missing features.
6. RTP (Real-Time Transport Protocol)

RTP is designed to handle real time traffic on the internet. It defines a standardized packet to deliver audio, video and other multimedia over the IP networks.
It is used for mainly for entertainment and communications in which mainly streaming is required.
RTP is used in those areas where the delivery of each and every packet is NOT of that much importance but the timing (delay) is more important. For example let’s consider streaming a video on youtube, where there is an acceptable limit under which the video should get buffered (depending on the bandwidth of the connection, if the bandwidth is higher than the video will get buffered faster otherwise it will take time accordingly).

RTP is used in conjunction with RTCP (Real-time Transport Control Protocol), which is used to monitor transmission statistics and Quality of Service (QoS).
The figure below shows the position of RTP which is between UDP and application layer.

RTP provides certain facilities like timestamping, sequencing and mixing.
Which Protocol to Use?
Now the question arises how to transfer data fast (that is with minimum possible delay) so for that purpose UDP protocol is used with some changes to it.
UDP is used because the transmission of data is fast using this protocol because of the reason that it doesn’t checks the acknowledgment (i.e whether the packet is received by the receiver or not)
Hence with some changes in the design itself UDP is used.

UDP Port
RTP is although a protocol itself but it is treated like application program so rather than encapsulating it directly with IP datagram it is encapsulated in a UDP user datagram.
The port number must be an even number because the odd number is used by its companion RTCP.

7. RTP Packet Format

Ver : This 2-bit field defines the version number. The current version is 2.

P. This 1-bit field, if set to 1, indicates the presence of padding at the end of the packet. In this case, the value of the last byte in the padding defines the length of the padding. Padding is the norm if a packet is encrypted. There is no padding if the value of the P field is 0.

X. This 1-bit field, if set to 1, indicates an extra extension header between the basic header and the data. There is no extra extension header if the value of this field is 0.

Contributor count. This 4-bit field indicates the number of contributors. Note that we can have a maximum of 15 contributors because a 4-bit field only allows a number between 0 and 15.

M. This 1-bit field is a marker used by the application to indicate, for example, the end of its data.

Payload type. This 7-bit field indicates the type of the payload.

Sequence number: This field is 16 bits in length. It is used to number the RTP packets. The sequence number of the first packet is chosen randomly; it is incremented by 1 for each subsequent packet. The sequence number is used by the receiver to detect lost or out of order packets.

Timestamp: This is a 32-bit field that indicates the time relationship between packets. The timestamp for the first packet is a random number. For each succeeding packet, the value is the sum of the preceding timestamp plus the time the first byte is produced (sampled). The value of the clock tick depends on the application. For example, audio applications normally generate chunks of 160 bytes; the clock tick for this application is 160. The timestamp for this application increases 160 for each RTP packet.

Synchronization source identifier: If there is only one source, this 32-bit field defines the source. However, if there are several sources, the mixer is the synchronization source and the other sources are contributors. The value of the source identifier is a random number chosen by the source. The protocol provides a strategy in case of conflict (two sources start with the same sequence number).

Contributor identifier: Each of these 32-bit identifiers (a maximum of 15) defines a source. When there is more than one source in a session, the mixer is the synchronization source and the remaining sources are the contributors

8. Real-time Transport Control Protocol

RTCP is used to control the flow and QoS and allow the recipient to send feedback. It partners with RTP in delivery and packaging of multimedia data, but does not transports any media streams itself. It consists of 5 type of messages.

Sender Report: The sender report is sent periodically by the active senders in a conference to report transmission and reception statistics for all RTP packets sent during the interval. The sender report includes an absolute timestamp. Which allows the receiver to synchronize messages.

Receiver Report: The receiver report is for passive participants, those that do not send RTP packets. The report informs the sender and other receivers about the quality of service.
Source Description Message: The source periodically sends a source description message to give additional information about itself. This information can be the name, e-mail address, telephone number, and address of the owner or controller of the source.
Bye Message: A source sends a bye message to shut down a stream. It allows the source to announce that it is leaving the conference. Although other sources can detect the absence of a source, this message is a direct announcement. It is also very useful to a mixer.
Application-Specific Message: The application-specific message is a packet for an application that wants to use new applications (not defined in the standard). It allows the definition of a new message type.

UDP Port
The UDP port chosen must be next number of port chosen for RTP.
9. VOICE OVER IP
It allows the communication between two parties over the packet-switched network instead of circuit switched network.
Two Protocols have been designed to handle this kind of communication: SIP and H.323
SIP (Session Initiation Protocol)
It is an application layer protocol that establishes, manages, and terminates a multimedia session (call). It can be used to create two-party, multiparty, or multicast sessions. SIP is designed to be independent of the underlying transport layer; it can run on either UDP, TCP, or SCTP.

The caller initializes a session with the INVITE message. After the callee answers the call, the caller sends an ACK message for confirmation. The BYE message terminates a session. The OPTIONS message queries a machine about its capabilities. The CANCEL message cancels an already started initialization process. The REGISTER message makes a connection when the callee is not available.

Simple Session

It includes establishing a session, communicating and terminating the session.
Tracking the Callee
SIP has a mechanism (similar to one in DNS) that finds the IP address of the terminal at which the callee is sitting. To perform this tracking, SIP uses the concept of registration. SIP defines some servers as registrars. At any moment a user is registered with at least one registrar server; this server knows the IP address of the callee.
The process is as follows

H.323
H.323 is a standard designed by ITU to allow telephones on the public telephone network to talk to computers (called terminals in H.323) connected to the Internet.

A gateway connects the Internet to the telephone network. The gatekeeper server on the local area network plays the role of the registrar server, as we discussed in the SIP protocol.

H.323 uses G.71 or G.723.1 for compression. It uses a protocol named H.245 which allows the parties to negotiate the compression method. Protocol Q.931 is used for establishing and terminating connections. Another protocol called H.225, or RAS (Registration/Administration/Status), is used for registration with the gatekeeper.

Operations

Reference:
1. Behrouz A. Forouzan, “Multimedia,” in TCP/IP Protocol Suite, 3rd ed. New Delhi, India , Tata McGraw-Hill Publishing Company Limited, 2007.

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