Signalling in voice over IP Networks

Voice signalling protocols have evolved, keeping with the prevalent move from circuit to packet switched networks. Standardization bodies have provided solutions for carrying voice traffic over packet networks while the main manufacturers are already providing products in workgroup, enterprise, or operator portfolio. This trend will accrue in next years due to the evolution of UMTS mobile networks to an “all-IP” environment. In this paper we present the various architectures that are proposed for signalling in VoIP, mainly: H.323, SIP and MGCP. We also include a brief summary about signalling in classical telephone networks and, at the end, we give some ideas about the proposed “all-IP” architectures in UMTS 3G mobile networks.Publicad

Investigating call control using MGCP in conjuction with SIP and H.323

Telephony used to mean using a telephone to call another telephone on the Public Switched Telephone Network (PSTN), and data networks were used purely to allow computers to communicate. However, with the advent of the Internet, telephony services have been extended to run on data networks. Telephone calls within the IP network are known as Voice over IP. These calls are carried by a number of protocols, with the most popular ones currently being Session Initiation Protocol (SIP) and H.323. Calls can be made from the IP network to the PSTN and vice versa through the use of a gateway. The gateway translates the packets from the IP network to circuits on the PSTN and vice versa to facilitate calls between the two networks. Gateways have evolved and are now split into two entities using the master/slave architecture. The master is an intelligent Media Gateway Controller (MGC) that handles the call control and signalling. The slave is a "dumb" Media Gateway (MG) that handles the translation of the media. The current gateway control protocols in use are Megaco/H.248, MGCP and Skinny. These protocols have proved themselves on the edge of the network. Furthermore, since they communicate with the call signalling VoIP protocols as well as the PSTN, they have to be the lingua franca between the two networks. Within the VoIP network, the numbers of call signalling protocols make it difficult to communicate with each other and to create services. This research investigates the use of Gateway Control Protocols as the lowest common denominator between the call signalling protocols SIP and H.323. More specifically, it uses MGCP to investigate service creation. It also considers the use of MGCP as a protocol translator between SIP and H.323. A service was created using MGCP to allow H.323 endpoints to send Short Message Service (SMS) messages. This service was then extended with minimal effort to SIP endpoints. This service investigated MGCP’s ability to handle call control from the H.323 and SIP endpoints. An MGC was then successfully used to perform as a protocol translator between SIP and H.323

Proprietary VoIP protocols of PBX manufacturers

Tato práce se zaměřuje na analýzu proprietárních Voice over IP protokolů. V první části je popsán princip a druhy technologie VoIP. V další části jsou popsáný proprietární protokoly Panasonic IP-PT od firmy Panasonic a Siemens CorNet-IP od firmy Siemens. Jsou zde uvedené protokoly, které jsou v těchto systémech využívány. Dále je popsána registrace terminálů k ústředně. Následně detailně dekódován průběh hovorů a jednotlivých funkcí terminálů včetně rozboru paketů. V závěru je provedeno srovnání s protokoly H.323 a MGCPThis thesis focuses on the analysis of proprietary Voice over IP protocols. The first part describes the principles and VoIP technology types. The next section describes proprietary protocols Panasonic IP-PT from Panasonic Corporation and Siemens CorNet-IP from Siemens Corporation. Protocols which are being used in these systems are listed here. Further there is being described the process of registration of terminals to central province. Subsequently calling process and particular terminal functions including packet analysis are being decoded in detail. In conclusion H.323 and MGCP protocols are being compared.

Protocolos de señalización para el transporte de voz sobre redes IP

En los últimos años, los protocolos de señalización para el servicio de transmisión de voz han experimentado una fuerte evolución junto con la tendencia a trasportar dicho tráfico desde las redes de conmutación de circuitos hacia las redes de conmutación de paquetes. Esta tendencia queda reflejada con la fuerte evolución de estándares en este ámbito y la aparición de productos en el mercado que cubren las necesidades de operadores, grandes empresas y PYMES [1] [2] [3]. Esta tendencia se verá incrementada durante los próximos 5 años debido a la evolución de las redes móviles basadas en tecnología UMTS hacia entornos “All-IP”. En este articulo se presentan las diferentes arquitecturas que están siendo propuestas para soportar la señalización de sistemas VoIP, debidas principalmente a los estándares H.323, SIP y MGCP, junto con una breve resumen de los mecanismos de señalización en redes telefónicas clásicas (SS7) y algunas ideas sobre la evolución hacia “ALL-IP” en redes móviles de 3G basadas en UMTS.Publicad

Service provisioning in two open-source SIP implementation, cinema and vocal

The distribution of real-time multimedia streams is seen nowadays as the next step forward for the Internet. One of the most obvious uses of such streams is to support telephony over the Internet, replacing and improving traditional telephony. This thesis investigates the development and deployment of services in two Internet telephony environments, namely CINEMA (Columbia InterNet Extensible Multimedia Architecture) and VOCAL (Vovida Open Communication Application Library), both based on the Session Initiation Protocol (SIP) and open-sourced. A classification of services is proposed, which divides services into two large groups: basic and advanced services. Basic services are services such as making point-to-point calls, registering with the server and making calls via the server. Any other service is considered an advanced service. Advanced services are defined by four categories: Call Related, Interactive, Internetworking and Hybrid. New services were implemented for the Call Related, Interactive and Internetworking categories. First, features involving call blocking, call screening and missed calls were implemented in the two environments in order to investigate Call-related services. Next, a notification feature was implemented in both environments in order to investigate Interactive services. Finally, a translator between MGCP and SIP was developed to investigate an Internetworking service in the VOCAL environment. The practical implementation of the new features just described was used to answer questions about the location of the services, as well as the level of required expertise and the ease or difficulty experienced in creating services in each of the two environments.KMBT_363Adobe Acrobat 9.54 Paper Capture Plug-i

Detecting and Mitigating Denial-of-Service Attacks on Voice over IP Networks

Voice over IP (VoIP) is more susceptible to Denial of Service attacks than traditional data traffic, due to the former's low tolerance to delay and jitter. We describe the design of our VoIP Vulnerability Assessment Tool (VVAT) with which we demonstrate vulnerabilities to DoS attacks inherent in many of the popular VoIP applications available today. In our threat model we assume an adversary who is not a network administrator, nor has direct control of the channel and key VoIP elements. His aim is to degrade his victim's QoS without giving away his presence by making his attack look like a normal network degradation. Even black-boxed, applications like Skype that use proprietary protocols show poor performance under specially crafted DoS attacks to its media stream. Finally we show how securing Skype relays not only preserves many of its useful features such as seamless traversal of firewalls but also protects its users from DoS attacks such as recording of conversations and disruption of voice quality. We also present our experiences using virtualization to protect VoIP applications from 'insider attacks'. Our contribution is two fold we: 1) Outline a threat model for VoIP, incorporating our attack models in an open-source network simulator/emulator allowing VoIP vendors to check their software for vulnerabilities in a controlled environment before releasing it. 2) We present two promising approaches for protecting the confidentiality, availability and authentication of VoIP Services

Protocolos para telefonia IP

Orientador: Nelson Luis Saldanha da fonsecaDissertação (mestrado) - Universidade Estadual de Campinas, Instituto de ComputaçãoResumo: A telefonia IP, também chamada de VoIP (Voice over IP), pode ser definida como qualquer aplicação telefônica usada em uma rede de comutação de pacotes de dados que utiliza o protocolo Internet Protocol (IP). Engloba novas aplicações que exploram a integração da comunicação de voz, imagens e de dados simultaneamente. Protocolos vêm sendo propostos para telefonia IP. No entanto, um grande desafio a ser transposto por estes protocolos é a garantia de qualidade de voz similar à da telefonia comutada por circuitos. Este trabalho apresenta os protocolos H.323, SIP, MGCP e Megaco/H.248 para telefonia IP, faz uma comparação destes protocolos e aborda fatores que afetam a Qualidade de Serviço (QoS) de telefonia IPAbstract: IP telephony can be defined as any telephonic application over the Internet Protocol and is one of the new applications that explore the integration of voice, image and data communication. Protocols have been proposed for IP telefony. However, one of the challenges in the IP telephony is to assure that the voice quality has similar quality of the one in circuit-switched telephony. This work presents the protocols H323, SIP, MGCP and MegacoIH.248 for IP telephony and compare them. It also describes the issues which impact the Quality of Service (QoS) in IP telephonyMestradoEngenharia de ComputaçãoMestre em Computaçã

Junos OS Security Configuration Guide

This preface provides the following guidelines for using the Junos OS Security Configuration Guide: • J Series and SRX Series Documentation and Release Notes on page xli • Objectives on page xlii • Audience on page xlii • Supported Routing Platforms on page xlii • Document Conventions on page xlii • Documentation Feedback on page xliv • Requesting Technical Support on page xliv Juniper Networks supports a technical book program to publish books by Juniper Networks engineers and subject matter experts with book publishers around the world. These books go beyond the technical documentation to explore the nuances of network architecture, deployment, and administration using the Junos operating system (Junos OS) and Juniper Networks devices. In addition, the Juniper Networks Technical Library, published in conjunction with O'Reilly Media, explores improving network security, reliability, and availability using Junos OS configuration techniques. All the books are for sale at technical bookstores and book outlets around the world. The current list can be viewed at http://www.juniper.net/books .Junos OS for SRX Series Services Gateways integrates the world-class network security and routing capabilities of Juniper Networks. Junos OS includes a wide range of packet-based filtering, class-of-service (CoS) classifiers, and traffic-shaping features as well as a rich, extensive set of flow-based security features including policies, screens, network address translation (NAT), and other flow-based services. Traffic that enters and exits services gateway is processed according to features you configure, such as packet filters, security policies, and screens. For example, the software can determine: • Whether the packet is allowed into the device • Which firewall screens to apply to the packet • The route the packet takes to reach its destination • Which CoS to apply to the packet, if any • Whether to apply NAT to translate the packet’s IP address • Whether the packet requires an Application Layer Gateway (ALG

Medição e caracterização de tráfego tempo-real em redes IP

Mestrado em Engenharia Electrónica e TelecomunicaçõesNos dias que correm os recursos financeiros e de tempo são cada vez mais escassos, é por isto que as novas tecnologias procuram economizar estes recursos. Um exemplo disto são os serviços tempo-real prestados pelas redes IP, como é o caso do VoIP e da videoconferência. Através de telefones VoIP é possível realizar chamadas telefónicas utilizando redes IP (por exemplo, Internet), desta forma aproveitam-se as redes de dados já existentes para transmitir voz. O VoIP oferece também mobilidade, pois um equipamento VoIP pode efectuar chamadas desde qualquer lugar, bastando que tenha acesso à Internet. Outra vantagem do VoIP é a interoperabilidade entre os diversos operadores (POTS e VoIP). A videoconferência permite também economizar recursos de tempo e financeiros, visto que através deste serviço o utilizador pode participar em eventos sem necessidade de se deslocar fisicamente. Além disto os equipamentos de videoconferência facultam a emissão de vídeo através da Internet, permitindo aos utilizadores assistir a eventos em tempo-real ou visualizar vídeos de eventos previamente gravados. Um dos objectivos desta dissertação é a medição e caracterização de tráfego com características de tempo-real em redes baseadas no protocolo IP, sendo um dos casos de estudos deste trabalho os serviços de VoIP e emissão de vídeo, prestados pela rede IP de investigação do IT de Aveiro. Para efectuar a medição e caracterização de tráfego encontram-se disponíveis diversas ferramentas, como é o caso do NTOP, TCPDump, TSTAT e TCPSTAT. Estas ferramentas são muito úteis, mas não são adequadas para efectuar uma medição distribuída, pois ao utilizar qualquer uma destas ferramentas para monitorizar diversos pontos de uma rede torna-se necessário a configuração e execução dos testes computador a computador, tornando esta tarefa muito complexa. De forma a tornar mais simples as medições distribuídas, a equipe de investigação do IT desenvolveu uma ferramenta de monitorização distribuída com uma arquitectura peer-to-peer hierárquica, trata-se do DTMS-P2P. Esta utiliza outras ferramentas existentes, por exemplo o TCPDump, mas acrescentando outras opções, que permitem uma medição distribuída sem necessidade de interagir nos diversos computadores cada vez que se deseja executar um teste ou consultar resultados. Esta ferramenta é também objecto de estudo desta dissertação, sendo contemplada no segundo caso de estudo. ABSTRACT: Nowadays the financial resources and time are increasingly scarce, it is because of this that the new technologies try to save these resources. An example of this is the real-time services provided by IP networks, such as VoIP and videoconferencing. With a VoIP phone is possible to make phone calls using IP networks (for example, Internet), by this way taking advantage of existing data networks for voice transmit. The VoIP also offers mobility, because the VoIP equipment can make calls from anywhere, having just an Internet access. Another advantage of VoIP is the interoperability between different operators (POTS and VoIP). The videoconferencing also can save financial and time resources, because through this service the user can participate in events without needing to physically move. Besides this the videoconferencing equipment provides video delivery over the Internet, allowing users to watch events in real-time or view events previously recorded. One of the objectives for this thesis is measurement and characterization of traffic with real-time characteristics in networks based on IP protocol, being one of the study cases of this work the VoIP services and video transmissions, provided by the investigation IP network on IT of Aveiro. To make the measurement and characterization of traffic, there are many tools available, such as NTOP, TCPDump, TSTAT and TCPSTAT. These tools are very useful, but are not suitable to make a distributed measurement, because when using any of these tools to monitor various points of a network, becomes necessary to setup and execute computer to computer tests, making this task very complex. In order to simplify the distributed measurements, the IT research team developed a tool for distributed monitoring with a peer-to-peer hierarchical architecture, it is the DTMS-P2P. This uses other tools available, such as TCPDump, but adding other options that allow a distributed measurement without needing to interact with the other computers each time you want to run a test or see results. This tool is also being studied in this thesis, being contemplated in the second study case