Archive for the ‘SIP’ Category

SIPit 22 begins today in New Hampshire…

Monday, April 14th, 2008

sipit.jpgWhat happens when a 100 or so SIP developers and engineers get together with all their gear? Well, starting today at the University of New Hampshire’s Interoperability Lab in Durham, NH, those developers will be participating in SIPit 22 and testing the interoperability of their solutions with those of everyone else. Sponsored by the SIP Forum and currently coordinated by Robert Sparks, the events provide a place for SIP implementors to test out how well their products work with others’. When you attend, you essentially arrange with others there to test your product with theirs and do so. The results are not published… it’s just a place for engineers/developers to go and work together on interop. Summaries of what occurred are released and you can get a sense of what goes one through the summary of SIPit 21 this past November in Beijing.

As you might expect, since I’m writing about this, one of our lead SIP developers is up there at SIPit 22 with our Prophecy product. As we’re working on some improvements in our SIP support that we’ll release in new versions of Prophecy over the next year or so, he’s looking forward to testing our new work and learning how well it works (or doesn’t work) with many other SIP implementations.

Given that SIP is composed of so many pieces and there are so many options in the way in which you can implement parts of SIP, events like SIPit that foster interoperability are really a key way to helping the industry grow. It’s great to see the SIP Forum continue to sponsor the SIPit events and we look forward to participating more in the future. Meanwhile, we’ll be very curious to see what results are brought home at the end of the week… :-)

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April 1 at IETF: SIPv4 to fix the problems of SIP and also Protocol Naming Rights

Tuesday, April 1st, 2008

ietflogo-2.jpgIn the IETF spirit that gave us RFC 1149 (IP over Avian Carriers) and RFC 3251 (Electricity over IP), Hadriel Kaplan and Bob Penfield brought out today an Internet-Draft titled “Session Initiation Protocol (SIP) Version 4.0: P2P2PSIP“. You can guess from the abstract how it is going to go:

This document defines a new and improved version of SIP, which tastes great and is less filling than the previous SIP. This draft updates all previous and future RFCs related to SIP in SIPPING, SIMPLE, MMUSIC, BEHAVE, and so on.

This is admittedly something perhaps only a “standards geek” (like me) would enjoy but the Introduction carries that spirit along:

SIP version 2.0, originally defined in RFC 2543 and updated by RFC 3261 [RFC3261], has become quite popular among the “in” crowd. It has, however, not been used in a way that people think it should be used, and has several problems caused by the growth in complexity of the protocol, ambiguity in usage, lack of security, and need for backwards compatibility. This draft makes no serious attempt to fix any of that. Instead, this draft is aimed at creating a new version of SIP, so that manufacturers can sell new equipment and software, to help the global economy. This in turn will increase tax revenue for governments, which eventually may lead to increased funds for feeding children. Therefore the ultimate goal of this draft is to feed starving children.

Thus, to accomplish the goal of feeding starving children by selling new equipment or software, a new SIP version is required which is not backwards compatible: SIP v4.0.

One may wonder why it isn’t numbered SIP v3.0 - the answer lies in market research. After the equivalent of hundreds of man-years of careful research (accomplished in 2 minutes of Google searching), we have determined that even-numbered versions of protocols have far greater chance of success than odd-numbered versions. For example, IPv4, BGP4, SNMPv2, H.323v2, and SIPv2 are all largely successful, while IPv5, BGP3, SNMPv3, and H.323v3 were not (SIPv3 did not even make it into a draft!). [Note: IPv6 does not count as purely even-numbered because it is actually 2 times 3, an odd number, whereas IPv4 is both even and the square of even numbers, which explains IPv6's relative failure thus far]

Of course, given my interest in security, I enjoyed this part:

This draft mechanism has no security issues, because it is normatively stated it MUST NOT have any. This is assured by using an extra “S” in “SIPSS”, and optionally using DTLS, IPSEC, VLANs, or whatever. From a security perspective, we should note that the extra “S” makes it either twice as secure as “SIPS”, or at least plus one. The “S” in “SIPS” from RFC 3261 was infinitely more secure than just “SIP”, because “SIP” had no trailing “S”; and as all children know, infinity plus one is bigger than infinity… thus “SIPSS” is definitely more secure than “SIPS”.

The whole document gave me quite a good laugh, including a number of the “references” that were slipped in among the valid references as well as FIRE and BRIMSTONE and the “poison pills” to avoid tampering by other standards organizations. Some parts of the document will admittedly be better appreciated if you have gone to recent IETF meetings or participated on the mailing lists to understand some of the meanings, but if you are familiar in any way with SIP you will probably find it amusing. Kudos to Hadriel Kaplan and Ben Penfield for a very creative piece of work.

Also today the IETF released RFC 5241, “Naming Rights in IETF Protocols”, offering a new way to fund the ongoing activities of the IETF. The rationale being that if companies will pay millions of dollars to put their names on stadiums, why not pay something to brand parts of protocols? :-)

Happy April 1st to all our readers!

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P2P (peer-to-peer) SIP - List of implementations now available

Monday, March 24th, 2008

p2psip.jpgWant to try out peer-to-peer SIP? (P2PSIP?) One of the areas of work within the IETF right now that intrigues me the most is the whole effort around “peer-to-peer” SIP, a.k.a. “P2PSIP”. The idea is that you could have “SIP without servers”, i.e. a range of SIP endpoints (hard phones, soft phones, etc.) that would learn of each other and communicate with each other using SIP over a P2P network (also referred to as a “P2P overlay” or sometimes just casually as a “P2P cloud”).

In another post at some point, I’ll write more about why P2PSIP is intriguing to me (and for Voxeo), but for the moment I thought I’d post that David Bryan, chair of the IETF’s P2PSIP Working Group, has very nicely put up a page listing existing implementations of P2PSIP technology. There are several open source implementations, as well as a commercial implementation from David’s own company, SIPeerior Technologies (he’s the CEO). All of these are implementations of exists drafts and so they will undoubtedly change as the drafts evolve and morph into RFCs over the next months and years ahead. Still, if you’d like to experiment and get a sense of what people are working on, the implementations are now out there!

P.S. David has also posted the P2PSIP presentations from IETF 71 earlier this month.

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And so IETF 71 draws to a close…

Friday, March 14th, 2008

ietflogo-2.jpgIt’s been a long and exhausting week… There are so many things I have wanted to write about… the RUCUS BOF… the MEDIACTRL session… the IPv6 experiment in the first plenary… the P2P video presentation in the technical plenary… the SIPPING and SIP sessions… the SPEERMINT and PEPPERMINT sessions… meeting the co-author of a draft I wrote who I had never met… the VERY lively P2PSIP session this morning… my challenges streaming video with hot laptops… cheese steaks and random strangers… the 100 Gbps network that Comcast brought in (yes, you read that correctly!)… so many sessions… so many great conversations… so many great people… so much great work going on… so many great stories to tell…

Alas, those stories will have to wait for late Sunday or Monday (or next week)… right now it’s time for me to head offline for some family time before I head out Sunday afternoon to Orlando for VoiceCon!

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My schedule next week in the long days of IETF-71…

Friday, March 7th, 2008

ietflogo-2.jpgOn Sunday night I head down to Philadelphia for the IETF-71 meeting for the whole week. It will be a crazy week full of discussions and conversations about all the various standards under development. The RUCUS BOF I’ve mentioned before will be on Monday as is the SIPPING Working Group. MEDIACTRL Working Group (of key interest to us here at Voxeo) is on Wednesday as is SPEERMINT and PEPPERMINT (Hey, it’s IETF, you have to have cute names!). Thursday brings SIP, BEHAVE, AVT and ENUM and Friday morning winds it all up with the P2PSIP working group.

Being who I am, I’ll pretty much sit in all of the “Realtime Applications and Infrastructure” (RAI) working groups as sometimes activity in one group turns out to have great relevance to work in other groups (or to work here at Voxeo). I’ll be online the Jabber chat rooms probably much of the whole time as well.

If you’ve never seen the full agenda for an IETF meeting, it’s pretty incredible (at least to me!). In any given timeslot there are typically eight simultaneous meetings of various working groups, BOFs, research groups, etc. This makes sense if you remember that the IETF is developing standards for pretty much all aspects of the Internet. While I usually never leave the world of RAI, there are groups dealing with security, DNS, email, IPv6, network routing, time (seriously!), host configuration and pretty much every other subject you can imagine relating to the Internet. Take a look!

And yes, the days do begin with a breakfast at 8am and meetings that go until 7pm (often with additional ad hoc meetings afterwards). The good news is that the breaks between sessions usually have food and drink to keep you recharged.

For those attending who wish to stalkfind me, here below is the agenda I think I’ll be following (subject to the fact that it can, of course, change). Like I said earlier, it’s pretty much all of the RAI area.


MONDAY, March 10, 2008
0800-0900 Continental Breakfast - Franklin Hall Foyer

0900-1130 Morning Session I

RAI mmusic Multiparty Multimedia Session Control WG

1300-1500 Afternoon Session I

RAI rucus Ruducing Unwanted Communications using SIP BOF

1520-1720 Afternoon Session II

RAI ecrit Emergency Context Resolution with Internet Technologies WG

1740-1950 Afternoon Session III

RAI sipping Session Initiation Proposal Investigation WG

TUESDAY, March 11, 2008
0800-0900 Continental Breakfast - Franklin Hall Foyer

0900-1130 Morning Session I
One of these:

IRTF rrg Routing Research Group
OPS v6ops IPv6 Operations WG
RAI geopriv Geographic Location/Privacy WG

1300-1500 Afternoon Session I

RAI bliss Basic Level of Interoperability for SIP Services WG

1520-1720 Afternoon Session II

RAI avt Audio/Video Transport WG

1740-1840 Afternoon Session III
One of these:

IRTF asrg Anti-Spam Research Group
RAI simple SIP for Instant Messaging and Presence Leveraging Extensions WG

1850-1950 Afternoon Session IV

RAI xcon Centralized Conferencing WG

WEDNESDAY, March 12, 2008
0800-0900 Continental Breakfast - Franklin Hall Foyer

0900-1130 Morning Session I

RAI mediactrl Media Server Control WG

1300-1500 Afternoon Session I

RAI speermint Session PEERing for Multimedia INTerconnect WG

1510-1610 Afternoon Session II

RAI peppermint Provisioning Extensions in Peering Registries for Multimedia INTerconnection BOF

1610-1700 PGP Session
(Yes, I’m one of those people who does actually go to PGP key signings.)

pgp PGP Key Signing

1700-1930 IETF Operations and Administration Plenary - Salon G/H

THURSDAY, March 13, 2008
0800-1700 IETF Registration - Franklin Hall Foyer

0800-0900 Continental Breakfast - Franklin Hall Foyer

0900-1130 Morning Session I

RAI sip Session Initiation Protocol WG

1300-1500 Afternoon Session I
One of these:

IRTF hiprg Host Identity Protocol
SEC saag Security Area Open Meeting
TSV behave Behavior Engineering for Hindrance Avoidance WG

1510-1610 Afternoon Session II
One of these:

RAI avt Audio/Video Transport WG
RAI enum Telephone Number Mapping WG

1700-1930 Technical Plenary - Salon G/H

FRIDAY, March 14, 2008

0800-0900 Continental Breakfast - Franklin Hall Foyer

0900-1130 Morning Session I

RAI p2psip Peer-to-Peer Session Initiation Protocol WG

RUCUS web page changed to a new URL

Monday, March 3rd, 2008

ietflogo-2.jpgAs I mentioned previously, the “RUCUS” BOF about voice spam at IETF 71 in Philadelphia is one of great interest to us. Unfortunately BOF co-chair Hannes Tschofenig ran into a problem with his domain and had to move the page to a new URL: http://www.shingou.info/bof-rucus.html

If you saved the URL or sent it on to someone, you’ll need to update to using the new URL. If you didn’t visit the RUCUS page before, please do check it out - and feel free to join the RUCUS mailing list. Of course, if you can, please do join us in person in Philadelphia!

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So what exactly is a “SIP trunk”, anyway? (One definition is proposed…)

Wednesday, February 20th, 2008

ietflogo.jpgOne of the ironies of the language we use in this space is that we all have been talking about “SIP trunks” for a few years now, but nowhere has there actually been a formal definition of what exactly a SIP trunk really is!

Leaping into the fray now is Jonathan Rosenberg, author of a zillion Internet-Drafts and multiple RFCs, with his new I-D titled, appropriately “What is a Session Initiation Protocol (SIP) Trunk Anyway?” Here is the abstract:

The term “Session Initiation Protocol (SIP) Trunk” has become almost commonplace amongst vendors and SIP providers. Even though the notion of a ‘trunk’ has a well defined meaning in circuit switched systems, it has never been defined for SIP. This document provides a formal definition for a SIP trunk, discusses its scope and applications, and establishes best practices for identification and security of SIP trunks.

The document makes for good reading even if you are not overly familiar with the concepts behind SIP trunks. Jonathan is looking for feedback and there will I’m sure be continued discussion on this topic.

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EComm2008 - Jonathan Christensen of Skype and the “unrealized” vision of SIP…

Saturday, February 9th, 2008

ecomm2008.jpgOver on the EComm2008 blog, Lee Dryburgh posted the transcript of a fascinating interview with Jonathan Christensen, general manager of audio and video at Skype. The interview is well-worth a read as Jonathan provides a preview of his upcoming keynote at EComm 2008 with his view of Internet-based communication and talks about advances they have made at Skype with regard to wideband audio and echo cancellation. I do definitely agree with his statement around the improvements they’ve made with echo cancellation on the Mac. Ever since upgrading to the latest Skype, I’ve made many calls with it from my MacBook Pro without any headset whatsoever and have been told the quality has been excellent (and it has been for me when I’m talking to other headset-free Skype users).

Much more relevant to this blog, though, were Jonathan’s statements regarding SIP. At the beginning Jonathan mentions how he originally got very excited by the vision of SIP and ran around stirring up interest at Microsoft where he worked then. But at the end of the interview, Lee asked Jonathan to elaborate on his earlier comments about SIP. This is what Jonathan said:

Yes, so just one clarification - we use SIP. Where, by comparison to the other operators, we are one of the largest SIP users in the world. All of our SkypeOut minutes and SkypeIn minutes traverse the PSTN via SIP interfaces, basically. So, we use it as an interop protocol where we need to.

I think that the vision of the early SIP founders has been largely unrealunrealized [See comments] in the SIP world. SIP is typically just used for these very mundane trunking applications, like the one that we have, or sending calls between two networks and it’s just calls. The vision of multi-modal communications and rich end points has largely failed within the same. I think that a big part of this is that they didn’t pragmatically just solve basic problems like NAT traversal, for example. They also evolved the specification to the point that it no longer had its lightweight appeal. So, we’ll see, SIP will continue to be [the] dominant protocol in terms of this sort of narrowly defined scenarios but I think that, when it comes to rich communications, you are going to see more of this fragmentation. You’re going to see some islands of providers who are just solving the problems. Just making it work for the user and not being religious about the protocol for example.

Has the vision of rich communication over SIP been “largely unrealized”? What do you think? Are his statements true? Or exaggerated?

FYI, if you are attending EComm 2008 you’ll have a chance to hear Jonathan Christensen’s keynote directly. And if you aren’t yet attending EComm 2008, why not? :-)

P.S. For the record, we, too, are huge users of SIP for our connections to/from the PSTN and also throughout our hosted Evolution platform as well as our on-premise Prophecy product. Developers on our hosted platform also get by default SIP *and* Skype dial-in numbers for their applications.

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Notes from the SIP Forum SIPconnect Compliance Workshop

Friday, January 25th, 2008

1B3DCB2E-8184-471F-878D-12C1E30C7FC6.jpgToday here at the Internet Telephony Expo in Miami Beach, Florida, the SIP Forum held a “SIPconnect Compliance Workshop” to help people understand the newly announced SIPconnect 1.0 specification. What follows are some notes about the session. There were about 40 people in attendance.

NOTE: I recorded the session and at some point the audio recording will be made available through the SIP Forum website.


The session began at 10:00am with SIP Forum Managing Director Marc Robins provided an overview of the SIP Forum, its activities and its members. There are now over 4,000 individual “Participant” members (membership is free) and 36 “Full” members who financially sponsor the SIP Forum. Marc also hinted at several major announcements coming up in the next weeks.

Marc next outlined the value proposition for SIPconnect. One of his main points was that “1st generation IP PBXs are dumbed down” in that they have to connect to the PSTN and can’t do direct peering. The ideal is really to connect directly into VoIP service providers. SIP is the industry standard for VoIP, but it’s difficult for people to understand which of the many pieces of SIP are relevant and necessary.

Marc stated that the industry needs an “industry-accepted interconnection method”. “SIPconnect” specifies a reference architecture - it specifies the minimum IETF and ITU specifications required to have successful interconnection between an IP-PBX and VoIP service provider. The point is really to be a “universal approach to SIP trunking”. Everyone who is certified as “SIPconnect compliant” has gone through an engineering exercise to ensure that they are truly interoperable.

Marc indicated that for SIPconnect delivers customer cost savings, enables transparent feature transport, optimizes quality of service and provides security. For IP-PBX manufacturers, Marc indicated that it can provide a competitive advantage, eliminate proprietary interfaces and generally a more seamless selling proposition for customers. For Service Providers, they get improved QoS and security, the ability to offer higher quality services for IP-PBXs and the ability to forge strong relationships with IP-PBX vendors and new relationships with distribution channels. Customers save money by not having to purchase a TDM gateway, improves voice quality by removing gateway latency and most importantly they get a foundation for future applications and services. For distributors and VARS they eliminate all the PSTN interconnection woes, they have the ability to manage QoS and also to move security issues from the customer premise into the service provider’s cloud.

Next up was Chris Gatch, the CTO of CBeyond, who provided an overview of the SIPconnect Compliance Process: What is the SIPconnect Compliant program? What does it cost? How do I join? How do I maintain my status in the program?

Steps to become SIPconnect Compliant:

  • (optional) Join the SIP Forum to get a reduction in the licensing fee ($2500/yr versus $5000/yr).

  • Download and complete the application.
  • Complete the Compliance Survey
  • Execute the Licensing and Compliance Agreement

Your application is then reviewed by a SIP Forum Certification Committee to determine compliance. Chris noted that they will work with folks because the goal is to help people to become compliant. To maintain compliance, you have to pay the annual $2500 licensing fee and keep up with the standards.

Chris provided some links and noted that the consolidated survey results are available that give some insight into how compliant products are. He noted that there are currently 7 companies who have certified 10 products. The two IP-PBX vendors who have certified are Digium and Avaya. Chris noted that it’s not about getting feeds but rather in driving interoperability and compatibility. It needs to be as meaningful as saying “FXS” or “PRI”.

Next up was Mark Enstrom from Broadsoft who discussed the “Lessons Learned” from companies as they became SIPconnect compliant. He spoke of information they gathered from informal conversations with companies that became SIPconnect compliant. Mark’s suggestions for service providers included:

  • Document your processes.

  • Standardize PBX configurations.
  • Provide configuration guides.
  • Provide an external interface for partner self-certification. (Example)

The question was raised by a participant of whether you could take a SIPconnect compliant IP-PBX and just connect it to a SIPconnect-compliant Service Provider. The answer is that this is the ideal to which the standard is a step. You still should need to do interoperability testing but it should be faster with SIPconnect-compliant products. The goal is to get to that point where it is as easy as connecting in a PRI.

For IP-PBX vendors, Mark suggested these guidelines:

  • Become SIPconnect compliant

  • Promote the program with service providers
  • Implement the DIGEST authentication method
    • TLS is required by SIPconnect (has become a general exception for most Compliant participants)
    • DIGEST is used in deployments
  • Implement optional REGISTER method (versus using static registrations)
    • Saves headaches in interop and deployment
    • Use master registration
    • Less configuration on the SBC
    • Reduces/eliminates downtime due to static registration address changes.

Mark then discussed issues around supporting fax and modem deployments, basically indicating that services providers today really need to explicitly test fax/modem deployments and document/support only a few configurations. Many service providers are still using separate interfaces for fax/modem traffic.

Mark moved into NAT and firewall issues. Service providers need to document what they support and train their customers and channels. Most firewalls are not SIP-aware. If you can use a SIP-aware firewall, you’ll be better off. Optionally, you can use port-forwarding or far-end NAT traversal if you understand the security issues.

Next Mark reminded service providers that they need to NOT forget back office integration. BSS/OSS integration needs to be factored into planning. Don’t forget to include billing systems: “If you can’t bill for it, it’s just a hobby!”

As far as the economics, the cost savings are very real with the elimination of PSTN gateways. Going direct with IP allows additional revenue opportunities, such as providing DN/DID services to smaller companies and delivering services to individual end users.

After a break, Chris Gatch came back to do a “deep dive” walking through the SIPconnect technical recommendation line-by-line with the 10 or so folks who remained. (And I stopped recording notes to focus on the spec.)

Chris later discussed some ideas around what SIPconnect 1.1 might focus on. Some of the possible areas of work include:

  • Update to use RFCs since the time of SIPconnect 1.0

  • Clarify DIGEST vs. TLS
  • Address “Off-Net Call Flows”
  • More specific recommendations around NAT and firewall issues
  • Provisioning Schema Standard
  • Redundancy/Recovery Use Cases
  • Re-visit requirements around media capabilities

Chris emphasized that these are only ideas about what might go into the SIPconnect 1.1 spec. The workgroup for SIPconnect 1.1 is only forming now, so the scope of the 1.1 work is yet to be defined.

The meeting concluded around 1:10pm with some final remarks by SIP Forum Managing Director Marc Robins encouraging people to become more involved with the SIP Forum.

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SIP Forum to host SIP Connect Compliance workshop at IT Expo this Friday in Miami

Tuesday, January 22nd, 2008

1B3DCB2E-8184-471F-878D-12C1E30C7FC6.jpgFor those of you attending the Internet Telephony Conference and Expo this week in Miami Beach, Floriday, the SIP Forum will be holding a SIPconnect Compliance Workshop on Friday, January 25th, from 10am-1pm. The workshop is free and the agenda is available. If you are at the show, please do come on by and learn about this initiative from the SIP Forum to help ensure interoperability for SIP trunking between service providers and IP-PBX systems. I’ll be there and it would be great to meet anyone reading this blog.

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