SDVoIP—A Software-Defined VoIP Framework For SIP And Dynamic QoS

Software-defined networking (SDN) allows for the decoupling of the control and data planes, enabling more programmability and a global view of the network. Previous research indicates that traditional applications recreated using SDN principles allow for more granularity and customization. In this research, we extend the insights behind SDN to develop a Voice over Internet Protocol (VoIP) framework with the objective to enhance traditional Session Initiation Protocol (SIP) operation and quality of service (QoS) approaches. The contributions of this research are 2-fold: first, an SIP control application is implemented, which communicates with an SDN controller to provide VoIP call registration and call routing capabilities, thereby eliminating the need for specialized SIP proxy hardware devices; second, a dynamic QoS application is developed that provides the ability to make network-wide QoS decisions based on real-time network measurements of latency, bandwidth and packet loss. Functional validation of the framework is performed to verify its operation. The experiment results indicate that the proposed framework allows for enhancements to traditional QoS implementations.

Modelo Sip Seguro para una Comunicación extremo a extremo sobre IPV6

Modelo Sip Seguro para una Comunicación extremo a extremo sobre IPV6 Sip Security Model for End to End Communication on IPv6 Ross M. Benites, José L Quiroz, Raúl Villafani INICTEL-UNI, Lima 41 DOI: https://doi.org/10.33017/RevECIPeru2011.0024/ RESUMEN Las implementaciones VoIP hoy en día se han incrementado considerablemente, sin embargo no entodos los escenarios se tiene en cuenta los mecanismos de seguridad adecuados. Este último punto es muy importante a considerar el día de hoy , sobre todo por el agotamiento de las direcciones IPV4 y el despliegue hacia IPV6 de muchos de los servicios, donde aparecerán nuevas amenazas a la seguridad que trataran de opacar el gran auge de la tecnología VoIP. Si bien IPv6 fue desarrollado para solucionar muchas vulnerabilidades en seguridad que actualmente se ven presentes en IPv4, el hecho es que no logra alcanzar aún estas metas según pruebas realizadas. El protocolo SIP, el actor principal de la tecnología VoIP , requiere de la implementación de mecanismos de seguridad . Los escenarios actuales requieren terminales de usuario de alto rendimiento y soporte para adaptarse a mecanismos de seguridad heterogéneos o asumir relaciones de confianza. Sin embargo debemos tener en cuenta que hay varias combinaciones de soluciones de seguridad que son proporcionados por usuarios finales y los servidores. En este trabajo se expone un modelo de seguridad aplicado a un escenario experimental VoIP sobre el Internet de Próxima Generación (IPv6), que utiliza la seguridad salto a salto y extremo a extremo. El escenario propuesto se encuentra sobre una red local con direcciones IPv6. Utiliza dos servidores Asterisk implementados bajo las mismas características que cumplen la función de SIP Proxy, y se encuentran conectados mediante un enlace troncal SIP – TLS. Se utilizan además dos terminales de usuario (teléfonos IP) provenientes de una marca comercial conocida, registrados cada uno mediante el protocolo SIP-TLS a cada servidor Asterisk. En trabajos anteriores, se han realizado varios estudios sobre el rendimiento del uso de VoIP sobre IPv4 e IPv6 comparando los resultados [1], evaluación de mecanismos de seguridad para mantener la autenticación de usuario, confidencialidad e integridad de la señalización y media de los mensajes VoIP sobre las redes IPv4 [2] y [3]. Este trabajo se esboza en un marco de seguridad, donde se presenta un escenario basado en una red VoIP en IPV6 utilizando TLS y SRTP. TLS es utilizado para la seguridad en el establecimiento de la sesión con mecanismos de autenticación salto a salto y SRTP (Secure Real Time Protocol) para la seguridad del establecimiento del stream de media. Nos enfocaremos en analizar y evaluar la seguridad de los mensajes en este escenario sobre el protocolo de transporte seguro (TLS). Descriptores: sip, tls ,ipv6, srtp. ABSTRACT VoIP deployments today have increased considerably, but not all the scenarios consider appropriate security mechanisms. This last point is very important to consider today, especially the depletion of IPv4 addresses and the deployment of many services IPv6, where will new security threats to try to overshadow the great technology boom VoIP. Although IPv6 was developed to solve many security vulnerabilities are currently present in IPv4, the fact is that still fails to achieve these goals by testing. The SIP protocol, the main actor of VoIP technology requires the implementation of security mechanisms. The current scenarios require high-end user performance and support to adapt to heterogeneous security mechanisms or assume trust relationships. But keep in mind that there are various combinations of security solutions that are provided by end users and servers. This paper presents a security model applied to an experimental scenario VoIP over Next Generation Internet (IPv6), which uses hop by hop security and end to end. The proposed scenario is on a local network with IPv6 addresses. Use two Asterisk servers implemented under the same characteristics that act as SIP Proxy, and are connected via SIP trunk - TLS. They also use two user terminals (IP phones) from a known trade mark registered by each SIP-TLS protocol for each server Asterisk. In previous work, there have been several studies on the performance of VoIP using IPv6 and IPv4 and comparing the results [1], evaluation of security mechanisms to support user authentication, confidentiality and integrity of the signaling and media messages VoIP over IPv4 networks [2] and [3]. This paper outlines a framework of security, which presents a scenario based on a VoIP network in IPv6 using TLS and SRTP. TLS is used for the security session establishment authentication mechanisms hop by hop and SRTP (Secure Real Time Protocol) for the safety of the establishment of media stream. We will focus on analyzing and evaluating the security of the messages in this scenario the secure transport protocol (TLS) . Keywords: sip, tls ,ipv6, srtp.

Performance Metrics for SIP-Based VoIP Applications Over DMO

The evaluation studies need to investigate a determined performance metrics to understand and evaluate the examined scenarios. SIP-based Voice over IP (VoIP) applications over MANET, which behaves in a way similar to Direct Mode of Operation (DMO) in mission Critical Communication Systems, have two main performance categories related to the Quality of Service (QoS). The main performance metrics that are considered for the evaluation processes in this research are the SIP end-to-end Performance metrics as defined by the RFC 6076. The main performance metrics are related to the registration, the call setup, and the call termination processes. In this research study, the SIP performance metrics are based on a single SIP proxy server. For voice data, the QoS evaluation is based on two methods: The Objective method and the Subjective method. The Objective method considers the traffic throughput, end-to-end delays, packet loss, and jitter, while the subjective method considers the Mean Opinion Score (MOS), which is mostly related to the end users' experiences during voice calls.

High Load Control Mechanism for SIP Servers

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.

Designation and Realization of the Interworking Gateway between IMS and PSTN

The interworking with the PSTN network is a network integration issues which must be solved. M2PA, M2UA and M3UA were compared and M3UA was selected because of its stronger flexibility. For the convenience of equipment expansion, IP switch was used in the hardware designation of PSTN Gateway. The software designation was based on independent entity. Socket communication mode which is loosely coupled was used between different entities. The software was divided into the SIP Proxy Module, the Signal Adapter Module, the Service Adapter Module, the Code and Decode Resource Control Module, and so on. The conversion of format, code, identification between SIP and No.7 signaling was realized. Multi-channel controller was used in the conversion of voice stream between the E1 interface circuit mode and RTP/RTCP mode. The interworking between PSTN and IMS was realized through the gateway and the old equipments can be used in the new network. This work is helpful to promote the network integration work.