Security in Peer-to-Peer SIP VoIP

VoIP (Voice over Internet Protocol) is one of the fastest growing technologies in the world. It is used by people all over the world for communication. But with the growing popularity of internet, security is one of the biggest concerns. It is important that the intruders are not able to sniff the packets that are transmitted over the internet through VoIP. Session Initiation Protocol (SIP) is the most popular and commonly used protocol of VoIP. Now days, companies like Skype are using Peer-to-Peer SIP VoIP for faster and better performance. Through this project I am improving an already existing Peer-to-Peer SIP VoIP called SOSIMPLE P2P VoIP by adding confidentiality in the protocol with the help of public key cryptography

Delivering Live Multimedia Streams to Mobile Hosts in a Wireless Internet with Multiple Content Aggregators

We consider the distribution of channels of live multimedia content (e.g., radio or TV broadcasts) via multiple content aggregators. In our work, an aggregator receives channels from content sources and redistributes them to a potentially large number of mobile hosts. Each aggregator can offer a channel in various configurations to cater for different wireless links, mobile hosts, and user preferences. As a result, a mobile host can generally choose from different configurations of the same channel offered by multiple alternative aggregators, which may be available through different interfaces (e.g., in a hotspot). A mobile host may need to handoff to another aggregator once it receives a channel. To prevent service disruption, a mobile host may for instance need to handoff to another aggregator when it leaves the subnets that make up its current aggregator�s service area (e.g., a hotspot or a cellular network).\ud In this paper, we present the design of a system that enables (multi-homed) mobile hosts to seamlessly handoff from one aggregator to another so that they can continue to receive a channel wherever they go. We concentrate on handoffs between aggregators as a result of a mobile host crossing a subnet boundary. As part of the system, we discuss a lightweight application-level protocol that enables mobile hosts to select the aggregator that provides the �best� configuration of a channel. The protocol comes into play when a mobile host begins to receive a channel and when it crosses a subnet boundary while receiving the channel. We show how our protocol can be implemented using the standard IETF session control and description protocols SIP and SDP. The implementation combines SIP and SDP�s offer-answer model in a novel way

DCCP Simultaneous-Open Technique to Facilitate NAT/Middlebox Traversal

https://datatracker.ietf.org/doc/rfc5595/Publisher PD

The Impact of TLS on SIP Server Performance

This report studies the performance impact of using TLS as a transport protocol for SIP servers. We evaluate the cost of TLS experimentally using a testbed with OpenSIPS, OpenSSL, and Linux running on an Intel-based server. We analyze TLS costs using application, library, and kernel profiling, and use the profiles to illustrate when and how different costs are incurred, such as bulk data encryption, public key encryption, private key decryption, and MAC-based verification. We show that using TLS can reduce performance by up to a factor of 20 compared to the typical case of SIP over UDP. The primary factor in determining performance is whether and how TLS connection establishment is performed, due to the heavy costs of RSA operations used for session negotiation. This depends both on how the SIP proxy is deployed (e.g., as an inbound or outbound proxy) and what TLS options are used (e.g., mutual authentication, session reuse). The cost of symmetric key operations such as AES or 3DES, in contrast, tends to be small. Network operators deploying SIP over TLS should attempt to maximize the persistence of secure connections, and will need to assess the server resources required. To aid them, we provide a measurement-driven cost model for use in provisioning SIP servers using TLS. Our cost model predicts performance within 15 percent on average

Evaluating and improving firewalls for ip-telephony environments

Firewalls are a well established security mechanism for providing access control and auditing at the borders between different administrative network domains. Their basic architecture, techniques and operation modes did not change fundamentally during the last years. On the other side new challenges emerge rapidly when new innovative application domains have to be supported. IP-Telephony applications are considered to have a huge economic potential in the near future. For their widespread acceptance and thereby their economic success they must cope with established security policies. Existing firewalls face immense problems here, if they - as it still happens quite often - try to handle the new challenges in a way they did with "traditional applications". As we will show in this paper, IP-Telephony applications differ from those in many aspects, which makes such an approach quite inadequate. After identifying and characterizing the problems we therefore describe and evaluate a more appropriate approach. The feasibility of our architecture will be shown. It forms the basis of a prototype implementation, that we are currently working on

Using an External DHT as a SIP Location Service

Peer-to-peer Internet telephony using the Session Initiation Protocol (P2P-SIP) can exhibit two different architectures: an existing P2P network can be used as a replacement for lookup and updates, or a P2P algorithm can be implemented using SIP messages. In this paper, we explore the first architecture using the OpenDHT service as an externally managed P2P network. We provide design details such as encryption and signing using pseudo-code and examples to provide P2P-SIP for various deployment components such as P2P client, proxy and adaptor, based on our implementation. The design can be used with other distributed hash tables (DHTs) also