VoIP researcher and software professional
San Francisco, California (USA)
+1-917-6216392 (Pacific Timezone)
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Kundan Singh VoIP researcher and software professional San Francisco, California (USA) +1-917-6216392 (Pacific Timezone) LinkedIn | Yahoo! | Facebook |
Resume Publications Talks Software |
They said it: Love is about giving ... giving happiness and care ... giving love, there is no question of asking love in return. Love is not a business. Its not whether you win or lose, but how you play the game, that counts! Work like you don't need the money. Dance like no one is watching, and love like you have never been hurt. When we walk to the edge of all the light we have and take the step into the darkness of the unknown, we must believe that one of two things will happen. There will be something solid for us to stand on or we will be taught to fly.
I like reading story books, poems, watching movies, listening to music, sketching, painting, playing cricket, pool and wasting time on computer.
An aqueous cascade bathe my ankles. I watched the relentless rhythms, the rise and fall of the sea, whispering from its depth untold secrets of a bygone era. I think of the past. Content in its flow. Happy in its history, safe in the arms of loved ones... long gone. The pristine clarity clouded. Things change. Memories remain forever!
I am a PhD graduate from the Computer Science department of Columbia University. My PhD advisor is Prof. Henning Schulzrinne. My research interest includes IP telephony scalability and reliability, peer-to-peer Internet telephony, multimedia collaboration. My PhD thesis title is Reliable, Scalable and Interoperable Internet Telephony
In the past I have done research in the field of IP telephony. I started with SIP-H.323 signaling translation during my masters, continued on unified messaing and conferencing issues, and finally looked at various performance and scalability issues for large scale IP telephony systems and peer-to-peer Internet telephony for my PhD thesis. IP telephony started with transfer of telephony calls over Internet. Beyond cost savings, there are many other benefits of IP telephony, e.g., integration with web and email, programming how to route your calls, integrating video and web-based interactive voice response systems. I have also looked into providing a multimedia collaboration environment that allows synchronous and asynchronous communications among various collaborators, e.g., students and instructor in a class or people in a study group. More recently I am looking at scalable and robust video communication on the Internet.
You can browse through various papers I have published, various talks I have given, and p2p-sip blog articles I have written to know more about my current and past research work. I do systems research. Which means I get to implement the new ideas in the form of software or student projects.
I have worked in industry to further enhance my IP telephony and video conferencing expertise. At Adobe I implemented prototype SIP and P2P-SIP systems to assist Flash Player and at TokBox I implemented web-based video telephony using Flash. I have been doing a few fun projects in Flash and Python as well in my 39 Peers P2P-SIP, the Internet videocity, and Flash-VideoIOprojects. To know more about my past and current projects please view my resume.
| Reliable, Scalable and Interoperable Internet Telephony | PhD thesis defense presentation, NY, 35 min | |
| Introduction to VoIP | Basic introduction to VoIP using SIP, 60 min | |
| Peer-to-peer Internet telephony using SIP | Telecom Italia visiting Columbia, NY, 20 min | |
| Peer-to-peer Internet telephony using SIP | NOSSDAV, Skamania, WA, 15 min | |
| SIP Server Scalability | IRT group meeting, 60 min | |
| Peer-to-peer Internet telephony using SIP | Panasonic Digital Networking Lab, Princeton, NJ, 70 min | |
| Reliable and scalable Internet telephony | Job Talk at Bell Labs, New Jersey, 1 hr | |
| Reliable and scalable Internet telephony | IRT group meeting, 35 min | |
| Peer-to-peer Internet telephony using SIP | NY metro area workshop, CUNY, New York. 20 min | |
| Peer-to-peer IP telephony | IRT group meeting, 45 min | |
| MobileNAT: Mobility across heterogeneous address spaces | Presented the work I did last summer at Bell Lab to IRT group, 1hr | |
| A survey of Internet infrastructure reliability | My PhD candidacy exam talk, 45 min | |
| Media services in CINEMA | At Intel/Dialogic facility, Morristown, NJ, 1hr 45min | |
| A survey of Internet routing reliability | IRT group meeting, 45 min | |
| Deploying IP telephony | Computer science department colloquium | |
| Introduction to the Session Initiation Protocol | NYSERtech at Albany | |
| Integrating VoiceXML with SIP services | NY metro area workshop at Columbia University. | |
| An overview of CINEMA implementation | IRT group meeting | |
| CINEMA: Columbia InterNet Extensible Multimedia Architecture | CNRC student presentation | |
| VoiceXML and Internet telephony | IRT group meeting | |
| Multimedia conferencing using SIP | IPtel 2001 workshop at Columbia University, NY | |
| Deploying IP telephony | NY metro area workshop, IBM, Hawthrone. | |
| ITU WG H.323/SIP | VON developers conference | |
| Multimedia communication applications | IRT group meeting | |
| Research status | ||
| ITU WG H.323/SIP | VON developers conference | |
| Voice mail system using SIP/RTSP | IRT group meeting, 30 min | |
| Overview of SIP-H.323 gateway | Engineering group at Sylantro | |
| Overview of H.323 and SIP-H.323 gateway | IRT group meeting, 90 min |
| Event notification in CINEMA | Talk by Salman Abdul Baset on CINEMA event notification to IRT lab. | |
| Integrating VoiceXML with SIP services | Xiaotao Wu's talk at ICC 2003 conference. | |
| Scaling SIP servers | Sankaran's talk in IRT group meeting. | |
| Towards junking the PBX: deploying IP telephony | Wenyu's talk at NOSSDAV 2001. | |
| Unified messaging using SIP and RTSP | Henning's talk at IPTS 2000 | |
| Interworking between SIP/SDP and H.323 | Henning's talk at IPtel 2000 |
| IP telephony demo | Overview of IP telephony and demonstration | |
| ACM demo | Slides for ACM research fair, 2003 | |
| New CINEMA demo | Latest slides for CINEMA demo | |
| BAE Demo | Slides for CINEMA demo to BAE | |
| Demo for Intel/dialogic | Slides for CINEMA demo to dialogic engineers. | |
| ACM demo | Slides for ACM research fair demo, 2002 | |
| CINEMA Architecture | Single slide showing the architecture. | |
| Project overview | Overview of the projects I am doing. | |
| Old CINEMA demo | Describes the various CINEMA components, the demo call flows and architecture. | |
| CATT poster | CATT 2001 poster. | |
| CINEMA Architecture | Describes the various CINEMA components and architecture. | |
| Research overview | Overview of the projects I am doing. | |
| Demo setup | Old CINEMA demo setup with ephone, sipd, sip323, etc. | |
| CATT poster | CATT 2000 poster. | |
| Libsip++ overview | Overview of LIBSIP++ module. | |
| SIPum overview | Overview of sipum module. | |
| SIPconf overview | Overview of sipconf module. | |
| SIP323 overview | Overview of sip323 module. | |
| Demo setup | Old CINEMA demo call flows. |
I worked on a number of pieces of software at Columbia Internet Real-Time Lab. Some of these software pieces were sold by a former startup company named SIPquest which was acquired by CounterPath. I am no longer associated with these software pieces, so please do not send me licensing questions. Feel free to browse through the documents and APIs of these software pieces. I am also looking for students interested in building open-source versions of these software at my projects site. Drop me a mail if you would like to work on this.
Quick Links: CINEMA,
sip323: translator,
sipum: voicemail,
rtspd: media server,
sipua: useragent,
sipconf: conference,
sipvxml: voicexml
API docs:
libsipapi (SIP lib),
libconf (mixer),
sipconf (conference),
sipum (voicemail),
sipvxml (VoiceXML),
libnat (NAT/firewall),
sippeer (P2P-over-SIP)
I have also worked on several other libraries in CINEMA such as conferencing (libconf), NAT/firewall traversal (libnat). The complete test-bed architecture is descibed in a technical report, found on my publications page. There are various other individual component publications, describing individual components in detail. Some of the slides for demonstration of these software can be found at my talks page. In the past I also built a Web based user agent hello2web as a prototype application. An old web page with documents can be found at helloweb.
I have worked on a few open source software pieces related to P2P-SIP and web-based video communication. Please visit my 39 peers P2P-SIP project page to know more about the P2P-SIP software. Please visit my Internet videocity project page to know more about the video communication software. I also mentor students in doing software research projects related to Internet multimedia communication. Please visit my student project page for details on how to join or contribute.
More recently I am in love with the Python programming language. I think Python and ActionScript are the most developer-efficient programming languages in the genre of general-purpose application development and user-interface programming, respectively. I also have a programming blog listing some fun questions for programmers.
If you would like to know more about my recent work experience, please read my resume and/or CV below.
Computer Networks, Internet real-time and multimedia systems, computer communication protocols, peer-to-peer networks, Internet audio and video telephony and conferencing, unified messaging, and scalable and reliable systems and networks. Current research focus is in building global scale peer-to-peer Internet telephony network.
6Connex: Leading the architecture and development of social networking and communication component for enterprise virtual events.
TokBox: I designed and implemented the flex based TokBox client for video conferencing. I also did several prototype implementations for PC to phone calling, shared media viewing, distributed server infrastructure for low latency and automatic fail over of video calls.
Adobe: I implemented a SIP stack and a P2P library in ActionScript and built several prototype Flash-based applications such as integrated SIP+XMPP communicator, click-to-call Flash component, browser extensions for Firefox and IE for PC to phone calling, and a P2P-SIP user agent. My P2P implementation is based on Bamboo DHT and incorporates authenticated data storage, secure transport and reliability.
Columbia: During the initial years, I wrote an object-oriented SIP user agent library in C++, using our underlying SIP transaction and parsing library. I developed other components such as unified messaging voice mail and answering machine server, multimedia conference server, interactive voice response server and SIP-H.323 signaling gateway. I wrote reusable object oriented modules for the conference library and media-streaming library. Later, I built scalability and reliability mechanism for SIP servers that provide PSTN-grade availability (five nines) and scalability (ten million BHCA), albeit at much lower cost. I also developed techniques and built systems for robust and scalable peer-to-peer Internet telephony without incurring any server maintenance cost.
Bell Labs: I worked on MobileNAT that provides IP mobility for devices in private address spaces. I wrote the client application that implements DHCP client and server, and the driver that traps and alters IP packets on Windows XP. I also wrote the server application that runs on the Linux router, implements DHCP server and alters the NAT mapping.
Motorola: In a team of two, I developed a complete H.323 video conferencing client for Windows using external components for Q.931 and media codecs. I also helped in various other ongoing projects such as H.323-H.324 gateway, H.320-based video conferencing and debugging tools for embedded systems.
Teaching Assistant, Advanced Internet Services (COMS E6181-1), Columbia University, Fall 2001, with 48 students enrolled in the class, and primary responsibility of evaluating assignments and programming projects, and interacting with the students regarding the course material. I continued as the TA of this course for the subsequent offering of this course over Columbia Video Network (a distance learning programme) for Summer 2002, Fall 2002, Spring 2003 and Summer 2003, with main responsibility being designing and grading evaluation assignments, programming projects and final exams for the enrolled students.
Project Mentoring: In the more than five years as a PhD student in the Internet Real Time Lab., I supervised many student projects such as active badges, event notification and scheduling system, screen sharing, floor control, file sharing, interworking between instant messaging and voice calls, phone announcement service, application level gateway for NAT and firewall traversal, email by phone, audio quality measurement for conferencing, location service for 911 calls in SIP proxy server, and integrating MPEG support in our media server. List of projects at http://myprojectguide.org
I have extensive programming experience in C, C++, Python, Java, Tcl, ActionScript and Perl. I have worked on both Unix and Windows platforms, as well as on real time OS. I am familiar with various tools such as MySQL, Apache, TomCat, gcc/make, VC++, CGI, servlet, Flex Builder, LAMP/WAMP and CVS. I have worked with various hardware and software tools such as Cisco router 2600 series, Cisco IP phone, Nortel MCS 5100 system, Intel/Dialogic IP telephony, MySQL replication, Vovida's SIP and TRIP stacks, DNS SRV and NAPTR, DHCP server and client. I have working knowledge of software process including CMM quality levels and software design models. I have also worked with Linux kernel programming and Windows driver programming.
I have extensive experience with various Internet protocols such as Session Initiation Protocol (SIP), Real-time Transport Protocol (RTP), Real Time Streaming Protocol (RTSP), Session Description Protocol (SDP), VoiceXML, Simple Object Access Protocol (SOAP), Extensible Messaging and Presence Protocol (XMPP), ITU-T recommendations H.323, H.225.0, cryptography,security protocols,wireless/mobility protocols such as Mobile IP and some intra-domain mobility protocols for fast handoff, IP-PSTN interworking for telephony and related protocols. I have worked extensively on server scalability and reliability, and peer-to-peer systems.
United States Patent 7,453,852, Method and system for mobility across heterogeneous address spaces, Buddhikot; Milind M. (Cliffwood, NJ), Hari; Adiseshu (Matawan, NJ), Miller; Scott C. (Freehold, NJ), Singh; Kundan Narendra (New York, NY), Lucent Technologies Inc. (Murray Hill, NJ) , Filed: July 14, 2003, Awarded: November 18, 2008. Also has international applications.
United States Patent 7,257,201, System and method for unified messaging in inter/intranet telephony, Singh; Kundan (New York, NY), Schulzrinne; Henning (New York, NY), The Trustees of Columbia University in the City Of New York (New York, NY), Filed: Aug 13, 2001, Awarded: Aug 14, 2007. Also has international applications.
United States Patent 7,266,091, System and method for conferencing in inter/intranet telephony, Singh; Kundan (New York, NY), Nair; Gautam (New York, NY), Schulzrinne; Henning (New York, NY), The Trustees of Columbia University in the City Of New York (New York, NY), Filed: Feb 28, 2002, Awarded: Sep 4, 2007. Also has international applications.
I am an Indian citizen and a U.S. permanent resident (green card holder).
Complete list of my academic publications including my PhD thesis as well as invited talks can be found on my web page at http://kundansingh.com. Please visit http://39peers.net for my open source P2P-SIP effort, http://p2p-sip.blogspot.com for my thoughts on P2P-SIP and http://code.google.com/p/videocity for my open source video communication project.
The public switched telephone network (PSTN) provides ubiquitous availability and very high scalability of more than a million busy hour call attempts per switch. If large carriers are to adopt Internet telephony, then Internet telephony servers should offer at least similar quantifiable guarantees for scalability and reliability using metrics such as call setup latency, server call handling capacity, busy hour call arrivals, mean-time between failures and mean-time to recover. This thesis presents a reliable, scalable and interoperable Internet telephony architecture for user registration, call routing, conferencing and unified messaging using commodity hardware. The results extend beyond Internet telephony to encompass multimedia communication in general.
The architecture presented in this thesis deals with two aspects: at least PSTN-grade reliability and scalability of the Internet telephony servers, and interoperable Internet telephony services such as conferencing and voice mail using existing protocols. We describe the architecture and implementation of our Session Initiation Protocol (SIP)-based enterprise Internet telephony architecture known as Columbia InterNet Extensible Multimedia Architecture (CINEMA). It consists of a SIP registration and proxy server, a multi-party conferencing server, a gateway for interworking SIP with ITU's H.323, an interactive voice response system and a multimedia mail server. CINEMA provides a distributed interoperable architecture for collaboration using synchronous communications like multimedia conferencing, instant messaging, shared web-browsing, and asynchronous communications like discussion forum, shared files, voice and video mails. It allows seamless integration with various communication means like telephone, IP phone, web and electronic mail.
We present two techniques for providing scalability and reliability in SIP: server redundancy and a novel peer-to-peer architecture. For the former, we use DNS-based load sharing among multiple distributed servers that use backend SQL databases to maintain user records. Our two-stage architecture scales linearly with the number of servers. For the latter, we propose a peer-to-peer Internet telephony architecture that supports basic user registration and call setup as well as advanced services such as offline message delivery, voice mail and multi-party conferencing using SIP. It interworks with server-based SIP infrastructures.