SIP COMPRESSION

The wired line network has been well studied and widely used for a long time. Most of its protocols are so successful that passed the test of time. There are many similar tasks in mobile and wired line environment, and we would like to achieve compatible, inter-working solutions. So it is a plausible idea to use the protocols of the wired line network in mobile environment too. However, the mobile and wired line environment differ significantly; mainly the bandwidth is different in the two networks. Although the difference is going to be smaller with the help of new generation of mobile networks, it will still remain significant. An acceptable solution is to compress these protocols. We have not found such a solution in the literature so our opinion is that this article is the first dealing with SIP compression. We have created a demonstration system, which connects two SIP user agents to each other and ensures the compression and decompression of the messages between them. In this article we show our development about adapting various compressing algorithms for SIP compression, and we evaluate them

Presence Traffic Optimization Techniques

With the growth of presence-based services, it is important to provision the network to support high traffic and load generated by presence services. Presence event distribution systems amplify a single incoming PUBLISH message into possibly numerous outgoing NOTIFY messages from the server. This can increase the network load on inter-domain links and can potentially disrupt other QoS-sensitive applications. In this document, we present existing as well as new techniques that can be used to reduce presence traffic both in inter-domain and intra-domain scenarios. Specifically, we propose two new techniques: sending common NOTIFY for multiple watchers and batched notifications. We also propose some generic heuristics that can be used to reduce network traffic due to presence

Supporting group mobility in mission-critical wireless networks for SIP-based applications

Diplomityössä tarkastellaan viiveherkkien SIP-sovellusten verkkoalueiden välistä ryhmäliikkuvuutta langattomissa, IEEE 802.11x -pohjaisissa IPv4/IPv6 verkkoympäristöissä. Nykyaikaisissa kriisinhallintatehtävissä reaaliaikaisen viestinnän merkitys on viime vuosina vahvasti korostunut. Tähän tarkoitukseen käytetyt viestintäjärjestelmät ovat olleet tavallisesti erittäin kalliita. Langattomien teknologioiden nopea kehitys on kuitenkin suunnannut mielenkiinnon edullisiin, kaupallisiin siviilipuolen valmisratkaisuihin. Pitkät yhteydensiirtoviiveet ovat tärkeä ongelma reaaliaikaliikenteen yhteydensiirron kannalta. VoIP-pohjaisen puheliikenteen on todettu kestävän enimmillään suuruusluokkaa 100 ms olevia viiveaikoja palvelunlaadun ratkaisevasti kärsimättä. Linkkitason yhteydensiirron ohella duplikaattiosoitteiden tarkistuksella DHCP-osoitteenhaun aikana ja SIP-yhteyden uudelleenmuodostuksella on saumattoman yhteydensiirron kannalta olennainen merkitys. Ryhmäliikkuvuus on saanut osakseen paljon huomiota ad hoc -verkkojen tutkimuksessa. Työssä tutkitaan mandollisesti saavutettavia hyötyjä, joita ryhmäliikkuvuusmalli pystyisi perinteiseen yhteydensiirtotapaan nähden tuomaan hierarkkisissa infrastruktuurisissa SIP-verkoissa. Sovellustason liikkuvuutta ja signaloinnin tehokkuutta tarkastellaan kaistankäytön ja tietoturvallisuuden näkökulmasta. Kokeellisessa osiossa pyritään mallintamaan ryhmäyhteydensiirtoja yksinkertaisessa, simuloidussa ympäristössä. Päätelmien tueksi yhteydensiirtojen suorituskykyä arvioidaan lisäksi numeerisella analyysilla.This thesis studies the provision of group mobility during inter-domain hand-offs for delay-sensitive SIP applications over wireless IPv4/IPv6 network environment, based on the IEEE 802.11x platform. In contemporary disaster relief operations, the role of real-time communications has been strongly escalating over the recent years. The communication systems used for these ends have been conventionally very expensive. The rapid evolution of wireless technologies has brought the focus of interest to the affordable Common-Off-the-Shelf civilian applications. Long latencies during hand-offs for real-time traffic are a very important problem. As the studies have pointed out, the VoIP-based voice traffic can withstand maximum approximate disruption times of 100 ms, without too high degradation in the quality of service. Along with the link-layer hand-off, the duplicate address detection procedure during DHCP address acquisition and the SIP connection re-establishment both have a major impact on the hand-off latency. The group mobility has gained high attention in the research of ad-hoc networks. The work studies the benefits that this scheme could possibly bring over the conventional hand-offs in hierarchical infrastructured SIP networks. Different approaches to application-level mobility and the signaling efficiency are examined from the viewpoint of bandwidth usage and network security. In the experimental part, group hand-offs are modeled in a simple, simulated environment. In addition, a numerical analysis is used to assess the hand-off performance to support the made conclusions

Analysis of Session Establishment Signaling Delay in IP Multimedia Subsystem

Abstract. This paper investigates and analyzes SIP delay in the session establishment signaling procedure in the IMS system. We investigate the delay for end-to-end link scenarios such as WiMAX-to-WiMAX, UMTS-to-UMTS, UMTS-to-WiMAX and vice versa. The analyses consider three types of delays: transmission delay, processing delay and queuing delay. The obtained results show that the main delay of session establishment signaling process is due to the processing delay. In addition, the lower channel rate in the UMTS network as well as IMS service rate has significant impact to the session establishment signaling delay

Managing ClientInitiated Connections

The Session Initiation Protocol (SIP) allows proxy servers to initiate TCP connections or to send asynchronous UDP datagrams to User Agents in order to deliver requests. However, in a large number of real deployments, many practical considerations, such as the existence of firewalls and Network Address Translators (NATs) or the use of TLS with server-provided certificates, prevent servers from connecting to User Agents in this way. This specification defines behaviors for User Agents, registrars, and proxy servers that allow requests to be delivered on existing connections established by the User Agent. It also defines keep-alive behaviors needed to keep NAT bindings open and specifies the usage of multiple connections from the User Agent to its registrar. Status of This Memo This document specifies an Internet standards track protocol for the Internet community, and requests discussion and suggestions for improvements. Please refer to the current edition of the "Internet Official Protocol Standards " (STD 1) for the standardization state and status of this protocol. Distribution of this memo is unlimited. Copyright Notice Copyright (c) 2009 IETF Trust and the persons identified as th

Optimizing IETF multimedia signaling protocols and architectures in 3GPP networks : an evolutionary approach

Signaling in Next Generation IP-based networks heavily relies in the family of multimedia signaling protocols defined by IETF. Two of these signaling protocols are RTSP and SIP, which are text-based, client-server, request-response signaling protocols aimed at enabling multimedia sessions over IP networks. RTSP was conceived to set up streaming sessions from a Content / Streaming Server to a Streaming Client, while SIP was conceived to set up media (e.g.: voice, video, chat, file sharing, …) sessions among users. However, their scope has evolved and expanded over time to cover virtually any type of content and media session. As mobile networks progressively evolved towards an IP-only (All-IP) concept, particularly in 4G and 5G networks, 3GPP had to select IP-based signaling protocols for core mobile services, as opposed to traditional SS7-based protocols used in the circuit-switched domain in use in 2G and 3G networks. In that context, rather than reinventing the wheel, 3GPP decided to leverage Internet protocols and the work carried on by the IETF. Hence, it was not surprise that when 3GPP defined the so-called Packet-switched Streaming Service (PSS) for real-time continuous media delivery, it selected RTSP as its signaling protocol and, more importantly, SIP was eventually selected as the core signaling protocol for all multimedia core services in the mobile (All-)IP domain. This 3GPP decision to use off-the-shelf IETF-standardized signaling protocols has been a key cornerstone for the future of All-IP fixed / mobile networks convergence and Next Generation Networks (NGN) in general. In this context, the main goal of our work has been analyzing how such general purpose IP multimedia signaling protocols are deployed and behave over 3GPP mobile networks. Effectively, usage of IP protocols is key to enable cross-vendor interoperability. On the other hand, due to the specific nature of the mobile domain, there are scenarios where it might be possible to leverage some additional “context” to enhance the performance of such protocols in the particular case of mobile networks. With this idea in mind, the bulk of this thesis work has consisted on analyzing and optimizing the performance of SIP and RTSP multimedia signaling protocols and defining optimized deployment architectures, with particular focus on the 3GPP PSS and the 3GPP Mission Critical Push-to-Talk (MCPTT) service. This work was preceded by a detailed analysis work of the performance of underlying IP, UDP and TCP protocol performance over 3GPP networks, which provided the best baseline for the future work around IP multimedia signaling protocols. Our contributions include the proposal of new optimizations to enhance multimedia streaming session setup procedures, detailed analysis and optimizations of a SIP-based Presence service and, finally, the definition of new use cases and optimized deployment architectures for the 3GPP MCPTT service. All this work has been published in the form of one book, three papers published in JCR cited International Journals, 5 articles published in International Conferences, one paper published in a National Conference and one awarded patent. This thesis work provides a detailed description of all contributions plus a comprehensive overview of their context, the guiding principles beneath all contributions, their applicability to different network deployment technologies (from 2.5G to 5G), a detailed overview of the related OMA and 3GPP architectures, services and design principles. Last but not least, the potential evolution of this research work into the 5G domain is also outlined as well.Els mecanismes de Senyalització en xarxes de nova generació es fonamenten en protocols de senyalització definits per IETF. En particular, SIP i RTSP són dos protocols extensibles basats en missatges de text i paradigma petició-resposta. RTSP va ser concebut per a establir sessions de streaming de continguts, mentre SIP va ser creat inicialment per a facilitar l’establiment de sessions multimèdia (veu, vídeo, xat, compartició) entre usuaris. Tot i així, el seu àmbit d’aplicació s’ha anat expandint i evolucionant fins a cobrir virtualment qualsevol tipus de contingut i sessió multimèdia. A mesura que les xarxes mòbils han anat evolucionant cap a un paradigma “All-IP”, particularment en xarxes 4G i 5G, 3GPP va seleccionar els protocols i arquitectures destinats a gestionar la senyalització dels serveis mòbils presents i futurs. En un moment determinat 3GPP decideix que, a diferència dels sistemes 2G i 3G que fan servir protocols basats en SS7, els sistemes de nova generació farien servir protocols estandarditzats per IETF. Quan 3GPP va començar a estandarditzar el servei de Streaming sobre xarxes mòbils PSS (Packet-switched Streaming Service) va escollir el protocol RTSP com a mecanisme de senyalització. Encara més significatiu, el protocol SIP va ser escollit com a mecanisme de senyalització per a IMS (IP Multimedia Subsystem), l’arquitectura de nova generació que substituirà la xarxa telefònica tradicional i permetrà el desplegament de nous serveis multimèdia. La decisió per part de 3GPP de seleccionar protocols estàndards definits per IETF ha representat una fita cabdal per a la convergència del sistemes All-IP fixes i mòbils, i per al desenvolupament de xarxes NGN (Next Generation Networks) en general. En aquest context, el nostre objectiu inicial ha estat analitzar com aquests protocols de senyalització multimèdia, dissenyats per a xarxes IP genèriques, es comporten sobre xarxes mòbils 3GPP. Efectivament, l’ús de protocols IP és fonamental de cara a facilitar la interoperabilitat de solucions diferents. Per altra banda, hi ha escenaris a on és possible aprofitar informació de “context” addicional per a millorar el comportament d’aquests protocols en al cas particular de xarxes mòbils. El cos principal del treball de la tesi ha consistit en l’anàlisi i optimització del rendiment dels protocols de senyalització multimèdia SIP i RTSP, i la definició d’arquitectures de desplegament, amb èmfasi en els serveis 3GPP PSS i 3GPP Mission Critical Push-to-Talk (MCPTT). Aquest treball ha estat precedit per una feina d’anàlisi detallada del comportament dels protocols IP, TCP i UDP sobre xarxes 3GPP, que va proporcionar els fonaments adequats per a la posterior tasca d’anàlisi de protocols de senyalització sobre xarxes mòbils. Les contribucions inclouen la proposta de noves optimitzacions per a millorar els procediments d’establiment de sessions de streaming multimèdia, l’anàlisi detallat i optimització del servei de Presència basat en SIP i la definició de nous casos d’ús i exemples de desplegament d’arquitectures optimitzades per al servei 3GPP MCPTT. Aquestes contribucions ha quedat reflectides en un llibre, tres articles publicats en Revistes Internacionals amb índex JCR, 5 articles publicats en Conferències Internacionals, un article publicat en Congrés Nacional i l’adjudicació d’una patent. La tesi proporciona una descripció detallada de totes les contribucions, així com un exhaustiu repàs del seu context, dels principis fonamentals subjacents a totes les contribucions, la seva aplicabilitat a diferents tipus de desplegaments de xarxa (des de 2.5G a 5G), així una presentació detallada de les arquitectures associades definides per organismes com OMA o 3GPP. Finalment també es presenta l’evolució potencial de la tasca de recerca cap a sistemes 5G.Postprint (published version