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StratoSIP - SIP at a Very High Level
Eric Cheung, Thomas Smith, Gregory Bond, Pamela Zave
IPTComm 2011,
2011.
[PDF]
[BIB]
ACM Copyright
(c) ACM, 2011. This is the author's version of the work. It is posted here by permission of ACM for your personal use. Not for redistribution. The definitive version was published in IPTComm 2011 , 2011-08-01.
{Voice-over-IP holds the promise of opening up the telecom domain to a vast community of developers. However, due to the nature of real-time multi-party communication and the complexity of the protocol, writing SIP applications that operate correctly and robustly is very challenging and requires in-depth knowledge. Even with standardized programming paradigms such as the Java-based SIP Servlet API, SIP application programming remains difficult.
StratoSIP [1] is a new domain-specific language designed to make the development task much easier and accessible to programmers with little knowledge of SIP. It will be released as open source software in 2011. As well, an applica- tion written using StratoSIP will be deployed in a mission-critical conferencing service in 2Q 2011. This demonstra- tion will show a typical StratoSIP development cycle using this conferencing application as a use case. The goal is to demonstrate how the high-level abstractions offered by StratoSIP make it very easy to program complex call control, and how the underlying StratoSIP runtime library ensures correct and robust operation even under protocol races.}
Getting SIP Endpoints and Network Call Control to Work Well Together
Eric Cheung, Thomas Smith
Principles, Systems and Applications of IP Telecommunications,
2011.
[PDF]
[BIB]
ACM Copyright
(c) ACM, 2011. This is the author's version of the work. It is posted here by permission of ACM for your personal use. Not for redistribution. The definitive version was published in Principles, Systems and Applications of IP Telecommunications , 2011-08-01.
{The Session Initiation Protocol (SIP) was designed as a protocol to establish multimedia over IP sessions between endpoints. Several protocol elements have been introduced to support common features such as call redirection and transfer performed at the endpoints. At the same time, real-world deployments have shown that intermediaries are commonly deployed in the network to provide advanced features. Conflicts between these protocol elements and intermediary application servers may lead to failure of operations and undesirable feature interactions. This paper examines the problems and proposes a general solution suite which includes suggested handling of these protocol elements and feature interaction management. The benefits include smooth interoperability of endpoint protocol elements and network-resident applications, and desirable overall service behavior.}
An Evaluation Framework for Highly Available and Scalable SIP Server Clusters
Gregory Bond, Eric Cheung, Thomas Smith, Jong Yul Kim
IPTComm 2011,
2011.
[PDF]
[BIB]
ACM Copyright
(c) ACM, 2011. This is the author's version of the work. It is posted here by permission of ACM for your personal use. Not for redistribution. The definitive version was published in IPTComm 2011 , 2011-08-01.
{SIP server clusters provide scalability and high availability for SIP applications. The nature of telecommunications services dictates requirements for these capabilities that generally exceed what is necessary in the domain of web applications. There are a number of clustering solutions available to address these requirements, but due to significant differences in architecture and the lack of common terminology, the solutions are difficult to compare. In this paper, we propose system requirements that are necessary to achieve scalability and high availability. We then propose an abstract cluster architecture containing four common architectural elements. Solutions from three different vendors are described in terms of these abstract elements. The solutions are compared based on an evaluation framework that includes a number of criteria, such as robustness to failures of different types and numbers, overhead required to manage failures, and scaling efficiency.}
Specification and evaluation of transparent behavior for SIP back-to-back user agents
Gregory Bond, Eric Cheung, Thomas Smith, Pamela Zave
4th International Conference on Principles, Systems and Applications of IP Telecommunications,
2010.
[PDF]
[BIB]
ACM Copyright
(c) ACM, 2010. This is the author's version of the work. It is posted here by permission of ACM for your personal use. Not for redistribution.
The definitive version was published in Proc. of the 4th International Conference on Principles, Systems, and Applications of IP , 2010-08-02.
{A back-to-back user agent (B2BUA) is a powerful mechanism for realizing
complex SIP applications. The ability to create, terminate, and modify SIP
dialogs allows the creation of arbitrarily complex services. However, B2BUAs
must be designed with care so as not to disrupt service interoperability. A
commonly-stated goal is for B2BUAs to be as {it transparent} as possible
while achieving its design goals. Though the notion of transparency is
intuitively appealing, it is difficult to define. To address this issue,
this paper proposes a definition of transparency and
presents a formal model of a transparent B2BUA to serve as the
specification of transparency. From this specification, we identify issues
with both the realizability and desirability of this behavior, and suggest
modifications to the original model. We evaluate the behavior of a number of
public B2BUA implementations via testing, using some novel techniques to
create test cases based on the formal models.}
A Novel Implementation of Very Large Teleconferences
Eric Cheung, Gerald Karam
IPTComm 2010,
2010.
[PDF]
[BIB]
ACM Copyright
(c) ACM, 2010. This is the author's version of the work. It is posted here by permission of ACM for your personal use. Not for redistribution.
The definitive version was published in ACM IPTComm 2010 , 2010-08-02.
{Certain teleconferencing applications must host very large number of participants that exceeds the capacity of a single mixing media server. Traditionally multiple media servers are connected in a cascading arrangement to meet the capacity requirement. This paper discusses the shortcomings of the traditional approach such as lower audio quality and unfairness of speaker selection. It then presents a novel approach that exploits the flexibility of Voice-over-IP and the Session Initiation Protocol to move participants between a main conference mixer and one or more media-distributing replicators as their role change between active talk-listen and listen-only. Benchmarking of an implementation of the replicators on general-purpose computers shows that large capacity can be achieved without specialized hardware. Moreover, it is shown how this approach can augment an already deployed teleconferencing system without modifying the existing telephony features, thereby illustrating the power of modularity and application composition in the SIP servlet environment.}
Experience with Modularity in an Advanced Teleconferencing Service Deployment
Eric Cheung, Thomas Smith
Proceedings of the Thirty-First International Conference on Software Engineering,
2009.
[BIB]
{In this paper, we describe our experience with the design of an advanced teleconferencing service under two different frameworks -- an early implementation of the Distributed Feature Composition architecture, and the SIP Servlet API. The usual design goals of software modularity for encapsulation and reuse are pursued. Interestingly, two very different designs resulted. This paper discusses the factors that influenced our design decisions. In particular, we examine the different characteristics of the two frameworks as well as the maturity of project requirements, and illustrate the ways in which these factors affect various mechanisms for achieving software modularity. We also aim to draw on this experience to propose a set of design guidelines for building modular, composable SIP Servlet applications for Voice over IP and converged services. Final camera-ready copy for conference }
Abstractions for Programming SIP Back-to-Back User Agents
Pamela Zave, Gregory Bond, Eric Cheung, Thomas Smith
Third International Conference on Principles, Systems and Applications of IP Telecommunications,
2009.
[BIB]
{In SIP services, {it back-to-back user agents (B2BUAs)} are powerful but difficult to program correctly. StratoSIP is a high-level, domain-specific language for programming SIP B2BUAs safely. This paper describes the four major abstractions on which the language is based. It explains how each abstraction is used in programming, and how it is implemented in SIP. Because the abstractions are derived from the Distributed Feature Composition (DFC) architecture, StratoSIP programs compose easily with each other at runtime. The implementation of StratoSIP runs in SIP Servlet containers. }
Automatic Mute Detection,
Tue Nov 13 16:12:22 EST 2012
A system and methodology for managing a mute function muting a connection to a conference call or other communication is disclosed. The system is implemented when the mute function is activated on a line connected to a conference call through a telecommunications network. The method includes receiving a sound, determining whether the sound meets predetermined criteria and if the sound meets predetermined criteria then performing a mute status operation.
Method And System For Providing Communication Control Functionality At A Remotely Located Site Using A Distributed Feature Architecture,
Tue Oct 23 16:12:07 EDT 2012
A method for providing communication control functionality at a remotely located site using a distributed feature architecture which provides communication features and data records to an Enhanced Network User (ENU). The ENU connects to a Distributed Feature Network (DFN) in order to gain access to communications features contained in the DFN. Based on the type of communication requested, one or more feature boxes are created in order to implement the requested communication.
Method And Apparatus For Generalized Third-Party Call Control In Session Initiation Protocol Networks,
Tue Jun 12 16:10:43 EDT 2012
In one embodiment, the present invention is a method and apparatus for generalized third party call control in session initiation protocol networks. In one embodiment, a method for controlling a media negotiation with one or more endpoints in a network includes determining, for each endpoint, a current state of a corresponding port on a third-party controller and transitioning the corresponding port to a new state in accordance with a finite state machine that tracks the state of the media negotiation.
Call Tracking Using Voice Quality Measurement Probe,
Tue Feb 16 15:03:33 EST 2010
A system for monitoring quality of calls in a network. A server receives information concerning calls which begin or end in a network and at the same time receives measurements of a quality parameter such as bandwidth. This measurement data is provided by a probe, so that the server is able to correlate calls which are placed through the network with the quality of service in the network at that time. When the quality falls below a desired level, the server notifies the customer or the network operator so that appropriate action may be taken.
Method And System For Providing Communication Control Functionality At A Remotely Located Site Using A Distributed Feature Architecture,
Tue Jul 28 16:07:39 EDT 2009
A method for providing communication control functionality at a remotely located site using a distributed feature architecture which provides communication features and data records to an Enhanced Network User (ENU). The ENU connects to a Distributed Feature Network (DFN) in order to gain access to communications features contained in the DFN. Based on the type of communication requested, one or more feature boxes are created in order to implement the requested communication.
System and method for text translations and annotation in an instant messaging session,
Tue Nov 11 18:13:14 EST 2008
A system and method for translating received input from a sender to recipient in an instant messaging dialog is disclosed. The method comprises receiving instant messaging input from a sender for recipient, wherein the instant messaging input comprises at least one subculture specific term. A category is identified the defines a difference between the sender and the recipient and the received instant messaging input is modified from the sender by generating an output associated with the least one subculture specific term and based on the identified category. Multiple recipients in a chat session may also each receive a translated or annotated message according to characteristics of each individual recipient.
Telecommunication network system and method in communication services using session initiation protocol,
Tue Jul 31 14:58:21 EDT 2007
An implementation of a voice-over-Internet protocol (VoIP) system for accomplishing two-way, three-way and conference calling between two or more parties is disclosed, in which new call features are readily adapted. The VoIP system is implemented in a Session Initiation Protocol (SIP) framework in which aspects of Distributed Feature Composition (DFC) architecture are modified and applied to overcome known limitations in the adaptability of existing VoIP frameworks.
Enhanced call feature service,
Tue Jan 16 18:11:48 EST 2007
The present invention provides a system and method for managing calls for a subscriber, including accessing a calling plan while the subscriber is in an existing active call with at least one other party. The call is placed or received by the subscriber at a first device. At least one service feature of the calling plan is invoked during the active call. This service feature includes an add callers feature for adding at least one additional party to the existing active call between the subscriber and the other party. The service feature additionally includes a transfer devices feature for transferring the existing active call to at least one additional device of the subscriber and hanging up on the first device by the subscriber to confirm the transfer of the call to the additional device.
Method And System For Providing Communication Control Functionality At A Remotely Located Site Using A Distributed Feature Architecture,
Tue Jun 14 18:10:24 EDT 2005
A method for providing communication control functionality at a remotely located site using a distributed feature architecture which provides communication features and data records to an Enhanced Network User (ENU). The ENU connects to a Distributed Feature Network (DFN) in order to gain access to communications features contained in the DFN. Based on the type of communication requested, one or more feature boxes are created in order to implement the requested communication.
Signaling/media separation for telecommunication network system and method,
Tue Aug 17 14:58:24 EDT 2004
It is an object of the present invention to provide a general-purpose media layer for a distributed feature system. In accordance with an embodiment of the present invention, a distributed feature system environment is disclosed in which paths carrying media, e.g. voice or text, are fully separated from signaling paths. A plurality of media processing components are utilized to establish media channels separate from signaling channels and wherein each media processing component is assigned to one or more feature components in the distributed feature system.
Method for placing a call intended for an enhanced network user on hold while the enhanced network user is unavailable to take the call using a distributed feature architecture,
Tue Sep 24 18:08:29 EDT 2002
A method for placing a call intended for an enhanced network user on hold is disclosed. A calling party is enabled to select the type of information which is provided to the calling party while the calling party is on hold by using a Distributed Feature Network (DFN) architecture. The DFN includes a plurality of feature boxes with each feature box being created for the purpose of enabling a particular communication feature. A call intended for one of a group of ENUs is received by the DFN and an estimated hold time is determined for the call. The hold time is communicated to a caller associated with the call and the caller is provided with a list of options for information to be received by the caller while the caller is on hold. The caller's selection of a hold option is received by the DFN. A feature box is created for providing the caller with the selected information option, and the call is connected to the created feature box. When the DFN determines that one of the group of ENUs is available, the call is rerouted from the created feature box to the available ENU.
IPTCOMM Best Paper Award, 2009.
For paper entitled, "Abstractions for Programming SIP Back-to-Back User Agents"
