(as of May 19, 2016)
TUESDAY, JUNE 21
7:00am-8:00am: BREAKFAST in Belmont Foyer
8:00am-2:30pm: SIPNOC 2016 SIP Training (located in Belmont Ballroom I and II)
Back by popular demand, the SIP Forum is offering a special SIP training workshop to the SIPNOC agenda. Once again, we've teamed up with Mark Lindsey, Sherwin Crown and the team at ECG to provide SIPNOC 2016 attendees relevant training on the SIP protocol, as well as provide insights and considerations related to a variety of technical implementations.
In contrast to video training and online courses available elsewhere, during the SIP Training workshop, you'll be working with experienced trainers who build, improve, and debug VoIP systems every day.
The SIPNOC 2016 SIP Training Workshop Topics Include:
- SIP Call Control. How do calls get started? How are they changed? How is the codec negotiated? How is incompatibility worked out?
- Session Border Controllers. When are these devices used? How does an SBC improve security? How does NAT interfere with normal VoIP? How does an SBC solve the NAT Traversal problem for VoIP? How is SIP Trunking and Carrier Peering Established? How can audio and video be improved by an SBC?
- VoIP Media. How is audio and video encoded for transmission? What are codecs, and how are they selected? What is RTP? What is a jitter buffer, and what problems can it solve? How does Network congestion and QoS affect the user experience?
- VoIP/IMS Network Components. What are the common equipment and terms you'll find used in VoIP/IMS networks? Which components are often combined? How do component failures affect ongoing phone calls?
- The Future for VoIP Networks. What are the open challenges in most VoIP networks? What are the trends coming from software and hardware vendors? How does SDN and NFV affect network growth? How do you prepare to support VoLTE, and IPv6?
An added bonus: All SIP Workshop training participants will receive a hardcover training manual with detailed notes and diagrams covering all of the covered topics.
12:30pm-1:30pm: LUNCH in Belmont III and IV
3:00pm-6:00pm: SIPNOC 2016 WebRTC Primer. Presented by Dr. Alan Johnston (located in Belmont Ballroom I and II)
New for 2016! A new WebRTC Primer, presented by SIP and WebRTC expert Dr. Alan Johnston, will provide you with the latest about WebRTC standards and deployments and real-world interoperability, and get you started in analyzing, coding, and troubleshooting WebRTC applications! Starting with the basics, we will introduce WebRTC and show you how to setup a development environment on your PC for node.js. You will learn WebRTC APIs such as getUserMedia and RTCPeerConnection on Chrome and Firefox browsers.
We will introduce media protocols such as SRTP, and NAT traversal protocols such as STUN, TURN, and ICE. You will learn how to analyze WebRTC packets using Wireshark and interpret WebRTC internal information from the browsers. We will step you through a simple signaling channel, client-side, and server-side programming. Key topics such as security including DTLS and privacy will also be covered.
7:00-10:00pm: SIPNOC 2016 Attendee Welcome Reception (includes food and drink) in Belmont Foyer
WEDNESDAY, JUNE 22
7:30am-8:30am: Breakfast in Belmont Foyer.
8:40am: General Session Begins in Belmont Ballroom I and II.
8:40am-8:45am: Welcome, Introduction and Housekeeping.
8:45am-9:15am: Session Border Controller in a NFV Cloud. Presented by Ashish Jain, GENBAND.
SBC’s are critical network elements to secure and manage real time IP communication sessions. SBC’s are typically deployed in an integrated architecture with signaling and media functions closely coupled in physical appliances (“the box”). However, the rapidly growing and changing nature of multimedia communications in the cloud is forcing SBC architecture to evolve to better meet the new demands imposed on the network.
Service providers and enterprises aim to leverage huge capital and operational savings by evolving SBCs to Network Function Virtualization (NFV) infrastructure, and enjoy the benefits of elastic scalability and better resiliency.
Today’s session explores the fundamental tenets of virtualization, NFV, and what you need to know to transition from a “box” paradigm to running next generation SBC software in a cloud. Other key topics will include virtualization vs NFV – what’s the difference?; How to treat session signaling and media in a virtualized world; and Media and transcoding processing in the virtualized world.
9:15am-9:45am: Striking a Balance for Real Time Communications. Presented by James Rafferty, Dialogic.
SIP and other forms of real time communications have dramatically changed how service providers build their networks. But such networks have special requirements and SIP is not just another IP protocol. Components such as load balancers can help service providers to meet their needs for enhanced scalability and resiliency, but should also address meet the challenges presented by real time communications.
In this presentation, we will review the following key points: - Unique Requirements of Real Time Networks; - How Load Balancers can help; - Why SIP is different and needs special handling; and the presentation of a Case Study of Load Balancer for a Financial Network.
9:45am-10:15am: Using SIP to Defeat Robocalling. Presented by Mark Lindsey, ECG.
When the phone rings, is it a human being or a robocall? The expensive PSTN once ensured calling was too expensive for practical robocalls, but the efficiencies of deregulation -- and VoIP -- have brought pre-recorded advertisements to every telephone.
While the US Federal Trade Commission (FTC) has sponsored competitions to combat robocalling, the VoIP protocol standards are silent the subject. This talk will review the leading robocall-prevention methods with focus on using and extending the VoIP protocols to combat robocalling attacks.
10:15am-10:45am: The Liberty Global Approach to Making SIP Work for Business. Presented by Theo van den Berg, Liberty Global and Andreas Granig, SIPwise.
When deploying a Global business communication solution together with Sipwise, Liberty Global faced a number of interoperability challenges to be tackled. IP Phones, Mobile Apps, IP PBXs and Trunk Gateway devices all need to be remotely configured in a secure and automated way to provide access to HPBX and trunking services.
This session will discuss mechanics and interoperability practices to provision a diverse endpoint line-up in a large, multi-national scale from both a vendor's and operator's point of view. Focus of this session is on the importance of standardization in combination with End 2 End interop / conformance testing.
10:45am-11:00am: Refreshment Break.
11:00am-11:30am: What We Learned Launching Primo (A Viral OTT Using SIP & WebRTC). Presented by Marc Abrams, Primo.
There are many successful commercial voice services based on SIP, but they are definitely not mobile-optimized and most are actually mobile-unfriendly. As an MVNO, our goal was to launch an over-the-top (OTT) service to extend our innovative Ultra Mobile calling plans to a much wider audience in the US and all over the world while leveraging our strength and experience in running a successful international retail and wholesale international termination network.
We wanted a solution that could integrate easily with and use our existing SIP infrastructure and that would be memory, battery and network efficient on the smartphones our customers were using, and that would be optimized for the fun, ease of use and instantly available experience our users have come to expect from us. Existing solutions fell far short, so we found some key partners and developed and launched our own.
We knew a lot about WebRTC but wondered about whether the technology was ready, and whether there were enough pieces available from vendors to actually consider a solution based on it. We were attracted to WebRTC as a technology, based on study and demos, but as we evaluated products from the vendor community we realized there were many functional gaps that needed to be bridged. However, if we could cover these gaps, with help from the vendor community, we would be able to create a differentiated solution that was a lot more than a theoretical exercise or a technology demo. And we learned some important lessons in the design, implementation, and support of a viral commercial OTT service. These lessons are the subject of the proposed presentation: Infrastructure sourcing – finding WebRTC products that work today and which can scale for the future; App development – IOS vs. Android; Offshore Development – setting requirements and managing to conclusion; and controlling initial launch to insure positive results.
11:30am-12:00pm: Header Compression in a Software Defined Network. Presented by William Flanagan, Flanagan Consulting.
Streaming media in the form of video dominates Internet traffic at times and is forecast to grow particularly quickly on mobile devices where bandwidth is limited on the Radio Access Network. Adding voice connections to packet networks is well underway and several organizations explicitly work toward dismantling the circuit-switched telephone network.
To date, packets on IP networks largely use version 4 addresses, but IPv6 inevitably will add many more bytes to each packet. For compressed voice, sending a sample of sound every 20 ms, the packet headers far exceed the payload yet carry little information that changes. The bandwidth consideration applies to video, though to a lesser extent as the payload of each packet can be larger. Software Defined Networks (SDNs) don't need the full stack of headers on every packet. The central controller can substitute very short flow IDs to guide payloads through a switched network, not only saving very significant bandwidth but also reducing latency at each switch. This presentation details the header compression ratios and estimates bandwidth reductions.
12:00pm-12:30pm: Trends in Hosted Voice and Operational Considerations for Service Providers. Presented by Dave Walters, BroadSoft.
Businesses are increasingly selecting hosted voice services. Understand how the growth of hosted voice impacts the delivery of existing services such as SIP trunking and commonly bundled services such as IP access and network services. Learn about the operational impact of delivering hosted voice services, especially with respect to traditional service demarcation and evolving service delivery models.
12:30pm-1:30pm: LUNCH in Belmont III & IV
1:30pm-2:15pm: SIPNOC 2016 Keynote Address: New Frontiers in Multimedia Communications. Presented by Henning Schulzrinne, Columbia University.
The SIP Forum is very proud to have Henning Schulzrinne return to keynote this year’s SIPNOC event. Dr. Schulzrinne is universally acclaimed as the Father of the SIP standard.
Dr. Henning will speak on the topic of "New Frontiers in Multimedia Communications". While SIP-based communications is still dominated by voice and conference-style video, new challenges and opportunities are emerging. He will address four related topics:
- Phone numbers are no longer the dominant communication identifiers, but no viable alternative has emerged. How should we find, identify and discover participants?
- Messaging systems using standards such as SIMPLE and XMPP have faded, replaced by closed systems both for consumer and enterprise use. Why did this happen and is there a role for a standards-based, open collaboration system?
- Communications services for people who are Deaf or hard of hearing are changing more than they have in the past decade. Real-time text, automated speech-to-text systems and partial replacement of relay services by direct consumer-to-enterprise American Sign Language communication offers new opportunities for better services and may impact what enterprise call center applications and smartphones are expected to be capable of.
- One of the most valuable Internet of Things applications is video, whether the video feed is streaming traffic data or watching a pet at home. How can we better integrate IOT with SIP-based applications?
As CTO of the FCC from 2012 to 2014, Schulzrinne guided the organization’s work on technology and engineering issues, together with the FCC’s Office of Engineering and Technology. He advised on matters across the agency to ensure that FCC policies are driving technological innovation, including serving as a resource to FCC Commissioners. Schulzrinne joined the FCC in 2010 as an Engineering Fellow.
Dr. Schulzrinne also serves as Julian Clarence Levi Professor of Mathematical Methods and Computer Science and Professor of Engineering at The Fu Foundation School of Engineering at Columbia University. Over the course of his career, he has published more than 250 journal and conference papers and more than 70 Internet Requests for Comment (RFCs) – working to shape the key protocols for enabling voice-over-IP (VoIP) and other multimedia applications, including the Session Initiation Protocol (SIP) which have become Internet Standards. His research interests include Internet multimedia systems, applied network engineering, wireless networks, security, quality of service, and performance evaluation.
2:15pm-2:45pm: Enhanced CNAM. Presented by Hala Mowafy, Ericsson.
Enhanced Calling Name (eCNAM) is an NGN call management service that overcomes the limitations of classic circuit switched CNAM. eCNAM delivers a full length name, up to 35 characters, and could provide additional information about the caller in a SIP Call-Info header. Additional information obtained from authoritative databases could assist subscribers in identifying unwanted communications, including robocalls.
2:45pm-3:00pm: Refreshment Break
3:00pm-3:30pm: Call Validation Display Framework. Presented by Richard Shockey, Shockey Consulting and SIP Forum.
The IETF STIR proposal envisions a method by which operators can assert the origin and validity of calls using E.164 numbers between service providers. There is a strong desire to take the Call Validation data from the terminating service and display it, in some useable manner, on SIP User Agents. This session will provide an overall view of some concepts for Call Validation Display, including the challenges, possible standardization efforts, possible integration with eCNAM, and cover some deployment scenarios.
3:30pm-4:45pm: SIPNOC 2016 NNI Super-Session. Moderator: Richard Shockey. Panelists include Martin Dolly, AT&T; Chris Wendt, Comcast; and John Barnhill, GENBAND.
The IP-Network to Network Interface (NNI) Joint Task Force is a cooperative effort between the Alliance for Telecommunications Industry Solutions (ATIS) and the SIP Forum that has been meeting regularly to define specifications to support SIP-based Service Provider to Service Provider IP Interconnection. The Task Force is comprised of telecommunications technical experts representing a range of telephony service providers and suppliers, both large and small, which serve both consumers and businesses.
The Task Force completed it's initial work on two Recommendations for the NNI interfaces in July of 2015: one on the NNI SIP Profile and another on NNI Interconnection routing. See this announcement for more information.
ATIS and the SIP Forum agreed to continue cooperation on several items. Current work has focused on s review of the existing IETF STIR proposals on Caller ID Verification and Anti-Spoofing techniques with a vision and supporting documentation on how it can be implemented in North American networks. In addition, it is developing a VoIP Security Whitepaper.
This session will present a comprehensive review of the IETF STIR proposal. What it is, how it works and current timetables for completion and implementation.
4:45pm-5:15pm: Unified Communications Services Threat Management. Presented by Amitava Mukherjee, RedShift Networks.
Global Carriers and Cloud Communication Providers are moving rapidly towards SIP environments. This shift in technology has created a dire need for a new class of security product that can comprehensively and proactively handle the sophisticated nature of today's threat paradigm and network anomalies on a massive scale. The stakeholders must understand that while SIP networks and applications present great promise, they also present unique security requirements that are different from conventional data applications. Due to the real-time nature of communications combined with the complex interconnect involving many entities, the overall network complexity and threat vectors exposure is alarming.
Most VoIP cyber security and fraud detection solutions deployed today are reactive (after the fact and by looking at CDR records) or by using data centric firewalls solutions. The new generation of VoIP Hackers are constantly redirecting attacks, leveraging different global SIP Bot networks to channel and masquerade existing defense systems that rely on blacklisting IPs, domains and/or SIP URIs, call numbers etc. This approach simply doesn’t work because there are just too many of them to blacklist. When a VoIP cyber attack or VoIP fraud event happens it’s often too late, the network is already attacked and compromised.
The bottom line is that attacks and intrusions are often the precursors to more sophisticated cybercrime such as espionage, Voicemail hacking, Toll Frauds, SPAM over Internet Telephony (SPIT) attacks, Eavesdropping, Targeted DOS attacks on SIP endpoints, SIP Bots, Robocalls, leakage of sensitive information, etc. What is needed is an algorithmic way to thwart attacks and that is based on deep context driven network analytics. This new class of solutions needs to detect any SIP based network anomalies that can result in network downtime. Network anomalies can be VoIP cyber attacks, fraud, misconfigurations or troubleshooting issues. A large global criminal enterprise is perpetrating these coordinated VoIP attacks that is resulting in major losses to global carriers.
5:15pm-5:45pm: IETF/Standards Update. Presented by Mary Barnes, MLB@Realtime Communications and SIP Forum.
This session provides an overview of current SIP and related standards activities underway in the IETF. Conferees will come away with insight into future network and endpoint functionality, as well as understand the process for evolving and developing SIP and other Real-time Applications and Infrastructure protocols.
7:00pm-10:00pm: SIPNOC 2016 Beer and Gear Networking Reception and Attendee Dinner in Belmont Foyer
7:30pm-8:30pm: BoF Time
8:30pm-9:30pm: BoF Time
THURSDAY, JUNE 23
7:30am-8:15am: Breakfast in Belmont Foyer
8:30am: Day 2 General Session Begins in Belmont Ballroom I and II.
8:30am-10:30am: New for 2016! IMS/VoLTE Training Workshop. Presented by Manpreet Singh, iBasis
This new workshop on IMS/VoLTE will cover the evolution of the LTE network and its evolution from 3G. Attendees will be provided with a background on doing voice over a 3G network and a typical 3G network topology, and the network topology of a LTE network including IMS.
VoLTE fundamentals will be covered, including SIP-IMS; Description of the IMS network elements; Description of the EPC core; Domestic call flows; Roaming call flows (including preconditions) including Local Breakout with HPMN Routing, Local breakout with VPMN routing, and S8 home routing; and a discussion of important SIP headers for volte signaling. Other areas covered include Charging models and Quality of Service implementation.
10:30am-10:45am: Refreshment Break
10:45am-11:15am: Special SIPNOC Keynote. Presented by Russ Housley, Vigil Security.
Mr. Housley is the Founder of Vigil Security, LLC, and he is coauthor of “Planning for PKI” and “Implementing Email and Security Tokens” published by John Wiley & Sons. He has over 30 years of communications and computer security experience. He served as the Security Area Director for the Internet Engineering Task Force (IETF) from 2003 to 2007, and then he served as Chair of the IETF from 2007 to 2013, and then he served as Chair of the Internet Architecture Board (IAB) from 2013 to 2015. He is presently serving as a member of the IAB and Chair of the IETF STIR Working Group. He is also serving on the IANA Stewardship Transition Coordination Group (ICG) and the ICANN IANA Transition Implementation Oversight Task Force (IOTF). His expertise is in security protocols, system engineering, system security architectures, and product definition. He is the author of the Cryptographic Message Syntax (CMS), the security foundation for S/MIME. He is one of the authors of the Internet X.509 Certificate Profile (RFC 5280), commonly called PKIX Part 1. He made significant contributions to IEEE 802.11 Wireless LAN security standards, particularly IEEE 802.11i. He is one of the authors of the SDNS Message Security Protocol (MSP), the security cornerstone of the U.S. Defense Message System (DMS).
11:15am-11:45am: SIP Marketplace Ecosystem Health Check. Presented by Bruce Page, Current Analysis.
This session complements the SIPNOC technical program with a review of the current and near-future state of the SIP market ecosystem.
Technical standards and interoperability testing are the foundation for SIP's further development as the basis for next-generation communications products and services.
This session will review the drivers and barriers to SIP adoption in carrier networks and customer-premises equipment and software applications; examine lessons learned from a number of successful SIP market-development initiatives; and consider the possibility that the communications market may be nearing a "tipping point" with regard to wide-scale adoption of SIP-based communications infrastructures.
12:15pm-1:15pm: LUNCH in Belmont III and IV.
1:15pm-2:15pm: Checking and Debugging Modern SIP Applications with SALSA Format and Tools. Presented by Vladimir Beloborodov, Mera Software Services.
Modern IP-based communications need to be secure for their users. While it is a requirement, extremely important for privacy and safety, it often makes testing and debugging of the signaling plane(s) in VoIP applications quite challenging for their developers. Sniffing packets in the middle with tools like Wireshark becomes more problematic, especially when it comes to troubleshooting problems in the field.
A viable alternative approach in this situation is dumping the captured (sent and received) signaling packets right at the client's level (or in the browser or its plug-in, for web / WebRTC applications). However, at the moment there is no common, standardised format that would be good for such type of activity, and it would be sufficiently succinct and yet flexible and expressive for modern and future needs of event logging and archiving of the signaling messages in different types of communication applications, including light, embedded, or web clients.
The proposed session – aimed at the BOF format – will present the newly proposed SALSA format ("Simple Application Log and Signaling Archive"). The SALSA specs were in development for about a year now, together with some sample tools for them – both the specs and the tools being planned for the open / open-source public release during March / early April. The preliminary results are very promising and are worth presenting them to a public audience now.
The session will start with a brief overview of the limitations and challenges related to using low-level dump formats, like PCAP, and tools like Wireshark, for modern VoIP / WebRTC development needs. Then it will give a review of SALSA and its design principles. After that, several live demos will be made – with a follow-up discussion and Q&A interaction with the audience. Questions and comments during the course of the presentation will be welcomed as well.
2:15pm-2:45pm: A Review of SIPconnect 2.0. Presented by Andrew Hutton, Unify.
The SIPconnect 2.0 Task Group has produced a SIP Forum SIPconnect 2.0 Technical Recommendation that defines a common set of implementation rules for implementers who desire to interface an IP PBX with a SIP-enabled service provider. The recommendation specifies which standards must be supported, provides precise guidance in the areas where the standards leave multiple options, supplements functionality gaps in existing protocols, and identifies a baseline set of features that should be common to an IP PBX to service provider interface.
The work takes into account of the role SBC’s have on both the enterprise IP-PBX side and the service provider side. At a high level, the interface to the service provider utilizes a combination of SIP signaling and media transport using RTP/SRTP. The interface to end-users of the IP PBX can be SIP or any other protocol capable of establishing a voice session (such as WebRTC, H.323, directly connected analog “black phone”, etc.)
2:45pm-3:15pm: It’s Live! - The SIPconnect Certification Testing Program Primer. Presented by Robert Kinder, Cox Communications; and Tim Carlin, University of New Hampshire Interoperability Laboratory.
This session will provide an overview of the recently launched SIPconnect 1.1 Certification Testing Program by the SIP Forum and the UNH-IOL. In addition, the session will provide a step-by-step review of the process prospective participating companies will encounter when they sign up for the program.
3:15pm-3:45pm: SIPNOC 2016 Capstone.
3:45pm: SIPNOC 2016 Concludes.