arXiv subjects

Henning Schulzrinne

Publications and source records attributed to Henning Schulzrinne.

24 records · Page 2Linked to original sources

arXiv2010

On TCP-based Session Initiation Protocol (SIP) Server Overload Control

The Session Initiation Protocol (SIP) server overload management has attracted interest since SIP is being widely deployed in the Next Generation Networks (NGN) as a core signaling protocol. Yet all existing SIP overload control work is focused on SIP-over-UDP, despite the fact that TCP is increasingly seen as the more viable choice of SIP transport. This paper answers the following questions: is the existing TCP flow control capable of handling the SIP overload problem? If not, why and how can we make it work? We provide a comprehensive explanation of the default SIP-over-TCP overload behavior through server instrumentation. We also propose and implement novel but simple overload control algorithms without any kernel or protocol level modification. Experimental evaluation shows that with our mechanism the overload performance improves from its original zero throughput to nearly full capacity. Our work leads to the important general insight that the traditional notion of TCP flow control alone is incapable of managing overload for time-critical session-based applications, which would be applicable not only to SIP, but also to a wide range of other common applications such as database servers.

cs.NI↗

arXiv2008

Session Initiation Protocol (SIP) Server Overload Control: Design and Evaluation

A Session Initiation Protocol (SIP) server may be overloaded by emergency-induced call volume, ``American Idol'' style flash crowd effects or denial of service attacks. The SIP server overload problem is interesting especially because the costs of serving or rejecting a SIP session can be similar. For this reason, the built-in SIP overload control mechanism based on generating rejection messages cannot prevent the server from entering congestion collapse under heavy load. The SIP overload problem calls for a pushback control solution in which the potentially overloaded receiving server may notify its upstream sending servers to have them send only the amount of load within the receiving server's processing capacity. The pushback framework can be achieved by either a rate-based feedback or a window-based feedback. The centerpiece of the feedback mechanism is the algorithm used to generate load regulation information. We propose three new window-based feedback algorithms and evaluate them together with two existing rate-based feedback algorithms. We compare the different algorithms in terms of the number of tuning parameters and performance under both steady and variable load. Furthermore, we identify two categories of fairness requirements for SIP overload control, namely, user-centric and provider-centric fairness. With the introduction of a new double-feed SIP overload control architecture, we show how the algorithms can meet those fairness criteria.

cs.NI↗

arXiv2008

Measurement and Evaluation of ENUM Server Performance

ENUM is a DNS-based protocol standard for mapping E.164 telephone numbers to Internet Uniform Resource Identifiers (URIs). It places unique requirements on the existing DNS infrastructure, such as data scalability, query throughput, response time, and database update rates. This paper measures and evaluates the performance of existing name server implementation as ENUM servers. We compared PowerDNS (PDNS), BIND and Navitas. Results show that BIND is not suitable for ENUM due to its poor scaling property. Both PDNS and Navitas can serve ENUM. However, Navitas turns out to be highly optimized and clearly outperforms PDNS in all aspects we have tested. We also instrumented the PDNS server to identify its performance bottleneck and investigated ways to improve it.

cs.PF↗

arXiv2008

A VoIP Privacy Mechanism and its Application in VoIP Peering for Voice Service Provider Topology and Identity Hiding

Voice Service Providers (VSPs) participating in VoIP peering frequently want to withhold their identity and related privacy-sensitive information from other parties during the VoIP communication. A number of existing documents on VoIP privacy exist, but most of them focus on end user privacy. By summarizing and extending existing work, we present a unified privacy mechanism for both VoIP users and service providers. We also show a case study on how VSPs can use this mechanism for identity and topology hiding in VoIP peering.

cs.NI↗

arXiv2007

Cooperation Between Stations in Wireless Networks

In a wireless network, mobile nodes (MNs) repeatedly perform tasks such as layer 2 (L2) handoff, layer 3 (L3) handoff and authentication. These tasks are critical, particularly for real-time applications such as VoIP. We propose a novel approach, namely Cooperative Roaming (CR), in which MNs can collaborate with each other and share useful information about the network in which they move. We show how we can achieve seamless L2 and L3 handoffs regardless of the authentication mechanism used and without any changes to either the infrastructure or the protocol. In particular, we provide a working implementation of CR and show how, with CR, MNs can achieve a total L2+L3 handoff time of less than 16 ms in an open network and of about 21 ms in an IEEE 802.11i network. We consider behaviors typical of IEEE 802.11 networks, although many of the concepts and problems addressed here apply to any kind of mobile network.

cs.NI↗

arXiv2004

An Analysis of the Skype Peer-to-Peer Internet Telephony Protocol

Skype is a peer-to-peer VoIP client developed by KaZaa in 2003. Skype claims that it can work almost seamlessly across NATs and firewalls and has better voice quality than the MSN and Yahoo IM applications. It encrypts calls end-to-end, and stores user information in a decentralized fashion. Skype also supports instant messaging and conferencing. This report analyzes key Skype functions such as login, NAT and firewall traversal, call establishment, media transfer, codecs, and conferencing under three different network setups. Analysis is performed by careful study of Skype network traffic.

cs.NI↗