Signaling Compression (SigComp)

Internet Telephony: Architecture and protocols—an IETF perspective

Chapter 16 - Video Communication Networks

Real Time streaming Protocol (RTSP) is the de facto standard for streaming and almost all the set up boxes and media players support it. RTSP is a very simple text based protocol for streaming. And this simplicity is one of the main reasons for the popularity of the protocol. We propose the use of RTSP along with Real Time Protocol (RTP) for Adaptive Forward Error Correction (AFEC) based streaming. The idea is to use the existing semantics of the protocol with minimal overloading to achieve the goal. The use of the RTP is solely to provide a mechanism for the packet sequence number, which is the main requirement to ensure reliability. Since, we will be using existing protocol semantics (overloading the existing semantics), we do not have the luxury of using Negative Acknowledgements (NACKs) or information rich packet headers. However, we can add the intelligence in the client/server to interpret the overloaded data and achieve the same results.

Internet Protocol (IP) telephony is an alternative to the traditional Public Switched Telephone Networks (PSTN), and the Session Initiation Protocol (SIP) is quickly becoming a popular signaling protocol for VoIP-based applications. SIP is a peer-to-peer multimedia signaling protocol standardized by the Internet Engineering Task Force (IETF), and it plays a vital role in providing IP telephony services through its use of the SIP Proxy Server (SPS), a software application that provides call routing services by parsing and forwarding all the incoming SIP packets in an IP telephony network. SIP Proxy Server Performance closely examines key aspects to the efficient design and implementation of SIP proxy server architecture. Together, a strong design and optimal implementation can enable significant enhancements to the performance characteristics of SPS. Since SPS performance can be characterized by the transaction states of each SIP session, the book analyzes an existing M/M/1-network performance model for SIP proxy servers in light of key performance benchmarks, such as the average response time for processing the SIP calls and the average number of SIP calls in the system. It also presents several other real-world industrial case studies to aid in further optimizations. This book is intended for researchers, practitioners and professionals interested in optimizing SIP proxy server performance. Professionals working on other VoIP solutions will also find the book valuable.

The Session Initiation Protocol (SIP) traditionally establishes and manages its sessions with centralized servers, which have been challenged by issues with TCP/IP networks right from its beginning, for instance, issues with NAT traversal or network congestions. On the other hand, there are peer-to-peer and overlay techniques that potentially can make SIP working better under various network conditions. Overlay networks can provide a good abstraction from the real network, thus hiding the presence of NATs. Some proposals on peer-to-peer SIP have been recently presented. While these proposals attempted to solve a subset of challenges faced by traditional SIP, such as removing the centralized SIP entities, there is to the best of our knowledge no single solution offering a comprehensive view of the peer-to-peer SIP architecture and operation. This paper presents a novel approach for a peer-to-peer SIP system, using overlay techniques for signalling and media transport in the Internet. This system is based on the Ambient Networks Service-Aware Transport Overlay (SATO) system. The proposed system replaces the traditional SIP proxy/registrar function with a distributed lookup mechanism, adding overlay functionality to the SIP signalling and to the RTP traffic. Moreover, different from previous proposals, our approach deliberately places media/packet relays into the SIP/RTP paths, which allows an efficient session management and media communication.

Multi-core platforms are becoming prevailing in telecom infrastructures, and many SIP(Session Initiation Protocol) enabled applications are using Java as the development language and runtime environment. It is important to understand the workload characteristics and performance issues of Java based SIP stack over multi-core platforms. In this paper, two RFC 3261 compliant Java based SIP stacks, the JAIN-SIP and a proprietary SIP stack are studied in depth. Especially we focused on the typical performance issues caused by either Java language features or the workload of SIP protocol, including scalability of SIP stack, resource contention issue and GC impact with SIP memory usage pattern. Some optimization techniques and their drawbacks are also discussed along with the performance evaluation. It turns out that due to the complex combination of SIP protocol semantics and Java language features, the performance of Java based SIP stack tends to be far from competitive on multicore platforms. Extensive optimization and certain improvements of program structure and object management policy may help, however, may, on the other hand, sacrifice some key values of the java language, e.g. easy development.

Many research activities have been provided to allow Session Initiation Protocol (SIP) based communication in Peer-to-Peer (P2P). We have contributed to these activities by proposing a scalable proxy topology building and routing in P2P SIP network where SIP location service is integrated in proxies that run over peer entities. Therefore as SIP cannot work efficiently through a NAT (Network Address Translation) device without using a specific NAT traversal mechanism, our proposed architecture needs an efficient NAT traversal technique which can work easily and make end-to-end communication effective in our framework by meeting our previously defining requirements. This paper provides a suitable technique to our P2P SIP framework for SIP signaling and media flow. Our NAT traversal solution proposes an implementation of a smart SIP registrar server which stores the client end device's public contact information (IP address and port) instead of its private address and maintains this public address in the NAT device table. This approach allows the proxy node to always send the SIP signaling message to the correct address by checking the Registrar server before each transmission. To enable media flow to be exchanged between the two SIP clients, we use a media relay located in the public network. The proxy server of the originator of the signaling message is allowed to replace suitably the private addresses in the message body by the public addresses of the media relay. We provide some relevant experimental results to show that the approach is feasible.

To start voice, image, instant messaging, and generally multimedia communication, session communication must begin between two participants. SIP (session initiation protocol) that is an application layer control induces management and terminates this kind of sessions. As far as the independence of SIP from transport layer protocols is concerned, SIP messages can be transferred on a variety of transport layer protocols including TCP or UDP. Mechanism of Retransmission that is embedded in SIP could compensate for the missing packet loss, in case of need. This mechanism is applied when SIP messages are transmitted on an unreliable transmission layer protocol like UDP. Also, while facing SIP proxy with overload, it could cause excessive filling of proxy queue, postpone increase of other contacts, and add to the amount of the proxy overload. In the present work, while using UDP as transport layer protocol, invite retransmission timer (T1) was appropriately regulated and SIP functionality was improved. Therefore, by proposing an adaptive timer of invite message retransmission, attempts were made to improve the time of session initiation and consequently improve the performance. Performance of the proposed SIP was implemented and evaluated by SIPP software in a real network environment and its accuracy and performance were demonstrated.

Nowadays, session initiation protocol (SIP) is the most important application layer protocol for multi-media to create, modify, and terminate the sessions. SIP is the platform of next generation networks. In this way, SIP should be able to respond to the needs of such a largely-used network. One of the major problems in SIP networks is overload. The basic mechanism embedded in SIP cannot appropriately resolve the due problem by purely rejecting requests, which may cause collapse in network. Many methods have been proposed to overcome overload in SIP, among which, multi-agent systems are new agent-based and increasingly growing approaches. Multi-agent system is a powerful tool to model and develop the complex large scale distributed systems. In SIP network, each agent may point out to a server which is in interaction with the others. As a result, a distributed and complex system is simply modeled. The due complexity can be reduced by holonic organization. In this paper, SIP network is divided into geographical areas in which each holon controls an area. The entire network is controlled in hierarchical structure of holons. Holonics structure can be dynamically changed based on the circumstances parameters and the status of the network during execution. Holons implement an end-to-end window-based method. This method offers an appropriate window size for edge servers to control load from the beginning of the network and prevent network overload. The simulation results confirm improved throughput, as well as better reduced call setup delay, rapid response to change and fairness.

With the evolution and penetration of the IP-based core network and the mobile access networks, more and more legacy services are being transferred to the converged and unified IP-based platform. For realizing this transfer, SIP (Session Initiation Protocol) has been adopted by standards bodies for creating, modifying, and terminating multimedia sessions. SIP with application-level mobility support can replace MIP (Mobile IP) to provide a FMC (Fixed Mobile Convergence) solution. Taking the important role of SIP into account, it is critical to fully resolve the SIP security issues including service authentication, confidentiality, and integrity protection on both signaling and data transmission. IPSec (IP Security) and the combination of TLS (Transport Layer Security) and SRTP (Secure Real-time Transport Protocol) are two typical SIP security solutions that can protect signaling and data transmission from various security attacks. However, these security solutions don't consider the issue of mobility support during the handover of mobile nodes among different access networks. This paper, therefore, proposes an effective SIP security solution that realizes SIP security with seamless mobility support across heterogeneous mobile networks. We compare the proposal to existing approaches and show that it better satisfies the SIP security requirements in terms of security, mobility support, and performance.

The Session Initiation Protocol (SIP) has gained momentum and is being widely used both in the Internet and Next Generation Telecommunications network as the core signaling protocol. SIP operation relies on SIP servers which are responsible for forwarding of SIP messages. It has been shown that the performance of SIP servers is largely degraded during overload conditions due to the built in message retransmission mechanism of SIP. This is an obstacle in wide deployment of SIP for a realistic situation. In this paper we propose a distributed and adaptive window based overload control algorithm, which controls the amount of calls that are forwarded to a downstream SIP server in an attempt to prevent it from being overloaded. Our scheme uses the call establishment delay as a measure of the amount of load on the downstream server and unlike other methods does not rely on explicit feedback. Using simulations we show that our proposed method performs better than a commonly used overload control algorithm and exhibits fast convergence and is stable.

To facilitate learning with video streaming, a transport protocol that can send video is needed, such as HTTP (Hyper Text Transfer Protocol) or RTSP (Real Time Streaming Protocol). Both protocols have their respective advantages and disadvantages when sent over a MANET network. One of them is that the throughput on RTSP is greater but the use of memory is more, this is inversely proportional to the RTSP protocol. In this study, the delay value in scheme 1 HTTP and RTSP protocols has an average delay of 23.225 seconds and 19.075 seconds. The throughput has an average value of 699.165 kbps and 802.656 kbps. Packet loss has an average value of 0.428% and 17.9%. For the use of transmit power nodes have an average of -33 dBm and -31.25 dBm. Memory usage has an average value of 108.46 Mb and 149.6 Mb.While in scheme 2 the delay values for HTTP and RTSP protocols are 28 seconds and 26.1 seconds on average. Throughput 1186,803 kbps and 1316,076 kbps. Packet loss has an average value of 0.0325% and 18.69%. On the use of transmit power nodes have an average of -34 dBm and -32 dBm. Memory usage has an average value of 107.525 Mb and 154.275 Mb.

Selective parsing to enhance SIP performance

Future telephony and multimedia systems will use the Session Initiation Protocol (SIP) for signaling purposes. SIP is a text-based protocol that imposes challenges for an efficient message processing. The ability of SIP entities to process SIP messages quickly is crucial for the performance of these networks, which often have strict timing requirements, e. g., to keep the call setup delays small. This paper studies the performance of SIP message processing in SIP proxies, focusing mainly on the impact of message parsing. We perform a detailed delay analysis for the widely used SIP Express Router (SER). Our measurements show that message parsing actually contributes significantly to a SIP proxy's processing efforts, and therefore confirm other existing studies. However, our results also show that the overall delay in high-performance SIP proxies is stronger affected by other factors, in particular the operating system.

Chapter 14 - Securing Multimedia Communications