Video surveillance systems currently are undergoing a transition where more and more traditional analog solutions are being replaced by digital solutions. Compared with an analog video surveillance system, a digital video surveillance offers much better flexibility in video content processing or data transmission. At the same time, it, also, have ability to implement advanced features such as motion detection, facial recognition and object tracking. Applying digital systems, makes the security system capable of transmitting video through the Internet, so we need to study the different methods of video streaming over the network. Streaming is the process of playing a file while it is still downloading. Streaming video is a sequence of “moving images” that are sent in compressed form in a way that it can start being proceed before it is completely received like video clips on a Web page.
Here, some of the network protocols used in video streaming are described. The focus is on the features of most important protocols in video surveillance including TCP, UDP and RTSP.
Protocols in streaming technology:
Protocols are the rules implemented for a particular technology, which in streaming technology are used to carry message packets, and communication takes place only through them. Some of the protocols used in streaming technology are described as follows:
SDP, standing for Session Description Protocol, used to describe multimedia sessions in a format understood by the participants over a network. The purpose of SDP is to convey information about media streams in multimedia sessions to help participants join or gather information of a particular session. In fact, SDP conveys information such as session name and purpose, times the session is active, codec format, media in the session, Information to receive those media (addresses, ports, formats and so on). A participant checks these information and takes the decision about joining a session.
SDP is aimed primarily for using in large WANs (Wide-Area Network) including the internet. However, SDP can also be utilized in proprietary LANs (Local Area Networks) and MANs (Metropolitan Area Networks).
Dynamic Host Configuration Protocol (DHCP) is a network protocol that enables a server to automatically assign a dynamic IP address to each device that connected to the network. By this assigning, a new device can be added to a network without the bother of manually assigning it a unique IP address. The introduction of DHCP eased the problems associated with manually assigning TCP/IP client addresses, resulting in flexibility and ease-of-use to network administrators.
DHCP is not a secure protocol, since no mechanism is built to allow clients and servers to authenticate each other. Both are vulnerable to deception, as one computer can pretend to be another.
Real-Time Transport Protocol (RTP) is an internet protocol standard to manage the real-time transmission of multimedia data over Ditch Cable unicast or multicast network services. In other words, RTP defines a standard packet format to deliver real-time audio and video over IP networks. RTP does not guarantee real-time delivery of data, but it provides mechanisms for the sending and receiving applications to support streaming data. It is utilized in conjunction with Real-Time Transport Control Protocol (RTCP) to ensure that monitor data delivery for large multicast networks is provided and Quality of Service (QOS) can be maintained. Monitoring is used to detect any packet loss and to compensate any delay jitter.
RTP is used extensively in communication and applications which involve streaming media such as telephony or video teleconference applications. The recent application of RTP is the introduction of VoIP (Voice over Internet Protocol) systems which are becoming very popular as alternatives to regular telephony circuits.
Real-Time Control Protocol (RTCP) is the control protocol that works in conjunction with RTP to monitor data delivery on large multicast network. Providing feedback on the quality of service being provided by RTP, is the RTCP’s primary function.
RTCP control packets are periodically transmitted by each participant in an RTP session to all other participants. It is important to point out that RTCP carries statistical and control data, while RTP delivers the data. RTCP statistics contain sender or receiver reports such as the number of bytes sent, packets sent, lost packets and round trip delay between endpoints. RTCP provides a way to correlate and synchronize different media streams that have come from the same sender.
The main protocol in streaming is Real Time Streaming Protocol (RTSP), which used to transmit stored or live media data over the IP network. It provides client controls for random access to the stream content. This application layer protocol is used to establish and control either a single or several time-synchronized streams of continuous media such as video and audio. RTSP servers use the Transport RTP in conjunction with RTCP, so that RTP acts as the transport protocol and RTCP will be applied for QOS (Quality of Service) analysis and also synchronization between video and audio streams. Consequently, RTSP can both control and deliver real-time content. The RTP and RTCP are independent of the underlying transport and network layers. In fact, RTSP is considered more than a protocol and provides a simple set of basic commands to control the video stream.
RSTP is based on the bandwidth available between the client and server so that breaks the large data into packet sized data. This, applied to live data feeds as well as stored. So, client software can play one packet, while decompressing the second packet and downloading the third media files. This enables the real-time file to be heard or viewed by the user immediately without downloading the entire media file and also without feeling a break between the data files.
Some features of the Real Time Streaming Protocol are listed as follows:
- RTSP is capable of presenting media streams from different multimedia servers.
- Controlling and delivering real time media between a media server and large numbers of media clients are feasible by RTSP.
- Firewall friendly: Both application and transport layer firewalls can be easily handled by means of protocol.
- RTSP provides on-demand access of multimedia items such as stored real time audio/video files, live real-time feeds, or stored non real time items.
- New parameters or methods can be easily added in the protocol, so it enables extension.
- There is appropriate control on the server. The server cannot stream to clients in any way such that the client cannot stop the streaming.
- Frame level accuracy makes protocol more suitable for media applications.
- RTSP allows interoperability between client-server multimedia products from multiple vendors.
Hypertext Transfer Protocol (HTTP), as an application-level protocol, is the set of rules to transfer files (text, graphic images, sound, video, and other multimedia files) on the web, so servers exchange information by using these rules. HTTP uses a server-client model in which the Web browser is client. When a user opens this Web browser, an HTTP command will be sent to the Web server. The browser uses HTTP, which is carried over TCP/IP to communicate to the server and retrieve Web content for the user.
It is worth mentioning that, HTTP is used for distributed, collaborative, hypermedia information system in addition to the context of World Wide Web.
The Real Time Messaging Protocol (RTMP) is used to transfer audio, video, and meta-data across a network. In fact, it is a system to deliver on-demand and live media to Adobe Flash applications which was developed by Adobe Systems. RTMP is a TCP-based protocol which maintains persistent connections and allows low latency communication. Splitting streams into fragments leads to delivering streams smoothly while transmitting much information. RTMP supports video in MP4 and FLV and audio in AAC and MP3.
Some advantages of RTMP include that it can do live streaming, allowing people to watch a video while it is being recorded. Also, it is capable of dynamic streaming, meaning that video quality adjusts automatically to bandwidth changes and seeking to later parts in a video is possible, which is particularly useful for longer videos. Players maintain the tiny buffer rather than downloading a video during playback, thus less bandwidth is used. RTMP streaming is able to skip forward to anywhere in a video at any point in time, so you can skip forward to what you want to see, without any unnecessary waiting. While with HTTP, only what is already in browser cache can be viewed. When RTMP is used as a protocol, host will need to have a dedicated server installed for RTMP.
However, RTMP has several disadvantages: due to streaming data to the player, the bandwidth of the connection must be larger than the data rate of the video, so if the connection drops for a couple of seconds, the stream will stutter. Also, since it uses different protocols and ports with HTTP, it is vulnerable to being blocked by firewalls. The biggest drawback is that RTMP only works in Flash and not in HTML5. Hence, it may be replaced by other streaming protocols with wider support.
Transmission Control Protocol (TCP) is a popular transport layer protocol which is connection-oriented and it supplies a reliable byte stream to the top layer, called as the application layer. TCP has a positive acknowledgments mechanism and also provides a mechanism for congestion avoidance to reduce the transmission rate when the network becomes overloaded. TCP guarantees that all packets arrive undamaged in the correct order, reordering out-of-order packets and/or asking a retransmit of lost packets.
To ensure the reliable data delivery over the network, the TCP employs window based transmission mechanism where the sender maintains a buffer, called a sliding window, of sent data to the receiver. A receiver acknowledges received data by sending acknowledgement (ACK) packets. If a sender receives an ACK packet for the data in its window, it removes that data from the window, because it has been successfully transmitted to the receiver. TCP employs this mechanism for controlling of flow, so that a receiver can tell the sender, when it cannot process the data at the arriving rate. This mechanism also informs the sender that how much buffering space is available at the receiver’s end, in order to avoid overfilling of receiver’s buffer window.