Requirements from Session Initiation Protocol (SIP) Session Border Control (SBC) Deployments

Bhatia, Medhavi; Camarillo, Gonzalo; Hawrylyshen, Alan; Penfield, Robert F.; Hautakorpi, Jani

doi:10.17487/rfc5853

Cited by 12 publications

(1 citation statement)

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“…The BRAINS entity is not expected to do SIP header harmonization (even if SIP is an IETF standard, there are several interpretations about the usage of certain information elements of header that can lead to interoperability issues between SIP peers), so in order to ensure that no interoperability issues appear when connecting to the external operators and institutes, another functional element is needed. The functions of this element, called a Session Border Controller (SBC), include hiding the internal network topology from external operators, provision of access control mechanisms, SIP header normalization, media flow handling and transcoding (change of codec between two domains) [6].…”

Section: Sip Trunking With the External Operatorsmentioning

confidence: 99%

Towards a 21st century telephone exchange at CERN

Valentin

Hesnaux

Sierra

et al. 2015

J. Phys.: Conf. Ser.

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Abstract. The advent of mobile telephony and Voice over IP (VoIP) has significantly impacted the traditional telephone exchange industry-to such an extent that private branch exchanges are likely to disappear completely in the near future. For large organisations, such as CERN, it is important to be able to smooth this transition by implementing new multimedia platforms that can protect past investments and the flexibility needed to securely interconnect emerging VoIP solutions and forthcoming developments such as Voice over LTE (VoLTE). We present the results of ongoing studies and tests at CERN of the latest technologies in this area. IntroductionCERN's current fixed telephony infrastructure, designed and implemented more than 20 years ago, is unsuccessfully struggling to adapt to modern communication needs, notably the support of multimedia capabilities and subscriber mobility. The reason for this is that it was designed as a Time Division Multiplexing (TDM) network, a technology that has its origins with the beginning of telephony. Key technology developments of the past 20 years, notably voice over IP (VoIP) and the signaling protocol SIP (Session Initiation Protocol) cannot be adopted easily or at all and our infrastructure has reached a dead-end in its evolution. This paper sets out our plan for revitalising the telephony infrastructure at CERN through the adoption of 21 st century technologies to replace those of the 19 th century. VoIP enables the transport of voice conversations, with the transmission of the original voice signal as a sequence of IP packets over today's widespread IP networks. The formal definition of the signaling protocol SIP [1], marked a milestone in the maturity of VoIP developments, as it enabled the transport not only of voice but also of video, instant messaging, file transfer and any kind or combination of real-time multimedia.SIP uses the underlying IP network to simplify enormously daily operation and subscriber management. With SIP, any user is able to connect a phone anywhere in the IP network and to change freely between different devices and even different technologies whilst keeping the same numberclear end-user advantages over the TDM model where a cabled phone line needs to be available. SIP also opens other communication channels for the corporate environment, where traditional Private Automatic Branch Exchanges (PABX) are being replaced progressively by Unified Communications (UC) solutions mimicking successful consumer-oriented applications as Skype. A second VoIP revolution is expected in the coming years with the worldwide rollout of fourth generation mobile networks, where the TDM domain will eventually disappear and all voice communications will be done using VoIP in the core networks where the IMS (IP Multimedia Subsystem) [2] built around the SIP protocol is a key enabler for VoLTE (Voice over LTE). A further VoIP traffic explosion will come from WebRTC (Real Time Communications on the Web) [3] that

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Section: Sip Trunking With the External Operatorsmentioning

confidence: 99%