Carol Davids

SIP APIs for voice and video communications on the web

Existing standard protocols for the web and Internet telephony fail to deliver real-time interactive communication from within a web browser. In particular, the client-server web protocol over reliable TCP is not always suitable for end-to-end low latency media path needed for interactive voice and video communication. To solve this, we compare the available platform options using the existing technologies such as modifying the web programming language and protocol, using an existing web browser plugin, and a separate host resident application that the web browser can talk to. We argue that using a separate application as an adaptor is a promising short term as well as long-term strategy for voice and video communications on the web. Our project aims at developing the open technology and sample implementations for web-based real-time voice and video communication applications. We describe the architecture of our project including (1) a RESTful web communication API over HTTP inspired by SIP message flows, (2) a web-friendly set of metadata for session description, and (3) an UDP-based end-to-end media path. All other telephony functions reside in the web application itself and/or in web feature servers. The adaptor approach allows us to easily add new voice and video codecs and NAT traversal technologies such as Host Identity Protocol. We want to make web-based communication accessible to millions of web developers, maximize the end user experience and security, and preserve the huge global investment in and experience from SIP systems while adhering to web standards and development tools as much as possible. We have created an open source prototype that allows you to freely use the conference application by directing a browser to the conference URL.

Dispatchable Indoor Location for Mobile Phones Calling for Emergency Services

This paper describes a proof of concept, working system that provides the indoor location of a caller who calls a Next Generation 9-1-1 Public Safety Answering Point using a smartphone. The paper includes descriptions of the system requirements and scope; The phone application that was developed; The location module that was developed; and the path taken by the call as it is delivered to the Public Safety Answering Point where the indoor location is displayed. The paper describes several architectures into which the modules of the service can be fit as well as plans for future development.

Real Time Communications in the Web: Current Achievements and Future Perspectives

The articles in this special section focus on the creation of real-time communication between web browsers via communication standards. These initiatives involve both the Internet Engineering Task Force (IETF) and the World Wide Web Consortium (W3C). The IETF has focused its work on the architecture and on-the-wire protocols, identifying the state information and the events that need to be available to developers, while the W3C worked on application-level JavaScript APIs to be offered to developers in order to allow them to easily implement interoperable real-time multimedia applications in the web. Back in 2011, the IETF chartered the “Real-Time Communication in WEB-browsers” (RTCWEB) Working Group, with the aim of defining an architecture and a complete suite of protocols for the support of real-time multimedia communications directly between browsers. The RTCWEB WG has since worked on key aspects like the overall communication infrastructure, the protocols and API (Application Programming Interface) requirements, the security model, the media formats (and related media codecs), as well as advanced functionality like congestion/flow control and interworking with legacy VoIP equipment. In parallel, the W3C has developed a set of APIs exposing functions like exploration and access to device capabilities, capture of media from local devices, encoding/processing of “media streams,” establishment of peer-to-peer connections between browsers (and web-enabled devices in general), decoding/processing of incoming media streams and delivery of such streams to the end-user in an HTML5-compliant fashion.

Benchmarking the session initiation protocol (SIP)

Measuring and comparing performance of an Internet multimedia signaling protocol across varying vendor implementations is a challenging task. Tests to measure the performance of the protocol as it is exhibited on a vendor device and conditions of the laboratory setup to run the tests have to be normalized such that no special favour is accorded to a particular vendor implementation. In this paper, we describe a performance benchmark to measure the performance of a device that includes a Session Initiation Protocol (SIP) proxy function. This benchmark is currently being standardized by the Internet Engineering Task Force (IETF). We implemented the algorithm that has been proposed in the IETF to measure the performance of a SIP server. We provide the test results of running the benchmark on Asterisk, the popular open source SIP private branch exchange (PBX).