Basic Telephony SIP End-to-End Performance Metrics

Morton, Al; Malas, Daryl

doi:10.17487/rfc6076

Cited by 34 publications

(17 citation statements)

References 3 publications

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“…Now considering this issue, through appropriate selection of Transmission protocol and also by selecting UDP as Transmission layer protocol, by regulating SIP Timers, we can provide conditions to increase the protocol effectiveness, in various network conditions and through considering the amount of Packet loss and Delay and the amount of Traffic load of SIP. As the session initiator can determine the Transmission protocol and the responder complies with him and making decision for this selection is done in each Nod of the SIP network [6], therefore, we can take measures in each point of the network considering the network conditions for selection [7][8][9]. In this case, the selection is done dynamically and without spending certain expense we can increase the transmission and as a result, the high efficiency [10,11].…”

Section: Related Workmentioning

confidence: 99%

The Overload Reduction in SIP Servers through Exact Regulation of the Retransmission Timer of the Invite Message

Montazerolghaem¹,

Yaghmaee²

2013

JCC

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To initiate voice, image, instant messaging and general multimedia communication, the Session communication must initiate between two participants. SIP (Session initiation protocol) is an application layer control, which task is creating management, and terminating this kind of Sessions. With regard to the independence of SIP from the Transport layer protocols, the SIP messages can be transferred on a variety of Transport layer protocols such as TCP or UDP. The mechanism of Retransmission, which has been embedded in SIP, is able to compensate the missing Packet loss, if needed. The application of this mechanism is when SIP messages are transmitted on an unreliable transmission layer protocol such as UDP. This mechanism, while facing with SIP proxy with overload, causes excessive filling of proxy queue, delays the increase of other contacts and adds the amount of the proxy overload. We in this article, while using UDP, as the Transport layer protocol, by regulating the Invite Retransmission Timer appropriately (T1), have improved the SIP functionality. Therefore, by proposing an Adaptive Timer of Invite message retransmission, we have tried to improve the time of Session initiation and as a result, improving the performance. The performance of the proposed SIP, by the SIP P software in a real network environment has been implemented and evaluated and its accuracy and performance has been demonstrated.

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Section: Related Workmentioning

confidence: 99%

The Overload Reduction in SIP Servers through Exact Regulation of the Retransmission Timer of the Invite Message

Montazerolghaem¹,

Yaghmaee²

2013

JCC

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“…The SIP protocol and SIP retransmission timers are defined in RFC 3261 [1], whereas SIP performance metrics for registration request delay are standardized in RFC 6076 [8]. C. Egger et.…”

Section: Related Workmentioning

confidence: 99%

“…Using metrics from [8], nearly 99% of all "401 Unauthorized" responses are sent within 1 ms and 98% of all "200 OK" within 2 ms of registration request arrival. Both values are well below the SIP retransmission timer T1 value of 500 ms, such that SIP server delay can be excluded as reason for timer T1 expiry.…”

Section: Sip Registrar Statisticsmentioning

confidence: 99%

Mobile SIP: An Empirical Study on SIP Retransmission Timers in HSPA 3G Networks

Fabini¹,

Hirschbichler²,

Kuthan³

et al. 2013

Lecture Notes in Computer Science

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Abstract. Mobile packet-switched voice must replace mobile circuitswitched voice latest with the large-scale deployment of LTE. However, short-term full-area LTE coverage is highly unlikely, handover to and interworking with 3G HSPA networks being a must. This raises the question of how the Session Initiation Protocol SIP performs in today's HSPA networks.Relying on live SIP provider monitoring data and active HSPA measurements, this paper demonstrates that default SIP retransmission timers are inappropriate for real-world deployments of packet switched voice over today's 2G and 3G networks including HSPA. Main reason for this deficiency is the reactive, demand-driven resource allocation strategy of HSPA which results in huge uplink delays. These delays eventually lead to a high number of unnecessary SIP message retransmissions which load networks and servers. SIP clients for mobile devices should therefore implement adaptive, user-configurable SIP retransmission timers and provide appropriate default values for these timers to enable seamless SIP operation in mobile cellular networks.

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“…Very little research work is done in the area of performance modelling of SIP proxy servers. Recently, lot of IP telephony industries is focusing on various SIP and SIP proxy performance metrics [17]. Rajagopal et al [11] analysed and proposed the IP Multimedia Services (IMS) network based on the SIP signalling delay and predicted performance trends of the network, which allowed them to choose parameter values optimally.…”

Section: Motivationmentioning

confidence: 99%

“…The SIP Registrar server is a network server which accepts/stores/serves registration requests and may interface with location services such as Light Weight Directory Access Protocol (LDAP), Common Object Request Broker Architecture (CORBA) and Open Database Connectivity (ODBC) database servers [9]. SIP proxy server accepts the session request sent by SIP UA, and queries the SIP registrar server to obtain the recipient UA's addressing and routing information [17]. SIP proxy server then forwards the session invitation directly to the recipient SIP UA if it is in the same domain or to another proxy server if the UA resides in a different domain.…”

Section: Introductionmentioning

confidence: 99%

A study of performance and scalability metrics of a SIP proxy server – a practical approach

Subramanian

Dutta

2011

Journal of Computer and System Sciences

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In recent years, Internet Protocol (IP) telephony has been a real alternative to the traditional Public Switched Telephone Networks (PSTN). IP telephony offers more flexibility in the implementation of new features and services. The Session Initiation Protocol (SIP) is becoming a popular signalling protocol for Voice over IP (VoIP) based applications. The SIP proxy server is a software application that provides call routing services by parsing and forwarding all the incoming SIP packets in an IP telephony network. The efficiency of this process can create large scale, highly reliable packet voice networks for service providers and enterprises. We established that the efficient design and implementation of the SIP proxy server architecture can enhance the performance characteristics of a SIP proxy server significantly. Since SIP proxy server performance can be characterised by its transaction states of each SIP session, we emulated the M/M/1 performance model of the SIP proxy server and studied some of the key performance benchmarks such as average response time to process the SIP calls, and mean number of SIP calls in the system. We showed its limitations, and then studied an alternative M/M/c based SIP proxy server performance model with enhanced performance model and studied additional key performance characteristics such as server utilisation, queue size and memory utilisation. Provided the comparative results between the predicted results with the experimental results conducted in a lab environment.

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Basic Telephony SIP End-to-End Performance Metrics

Cited by 34 publications

References 3 publications

The Overload Reduction in SIP Servers through Exact Regulation of the Retransmission Timer of the Invite Message

The Overload Reduction in SIP Servers through Exact Regulation of the Retransmission Timer of the Invite Message

Mobile SIP: An Empirical Study on SIP Retransmission Timers in HSPA 3G Networks

A study of performance and scalability metrics of a SIP proxy server – a practical approach

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