Speaking of Standards

Do “intrastate” versus “interstate” billing rates make any sense whatsoever in the world of SIP where geography ceases to be relevant?

There is a fundamental disconnect today between the legacy world of PSTN billing practices and this new world of SIP-based interconnection that we are all building. The Internet ignores geography. We can connect to any random points in the cloud, regardless of where they are physically located.

But until the day when we have truly built the IP-interconnect and can all be bypassing the PSTN, we do have to deal with the PSTN… and all the legacy tariff/billing infrastructure that goes with it. How do you mesh the two worlds?

Consider this case within the US:

• A company in Massachusetts with phone number 617-555-1111 calls a customer
  also in Massachusetts with phone number 617-555-2222.

• The Caller ID presented to the receiving party is 617-555-1111.

• The company calling is billed the intrastate rate because the call appears
  to have originated and terminated within Massachusetts. For this
  example, and purely picking a number out of the air, let’s say the call at intrastate rates would have cost $5.

• However, the company in MA was actually using a hosted telephony service
  and while the call appears to have originated from Massachusetts based
  on the CallerID presented to the user (and transmitted as “6175551111@sipgw.carrier.com” in the SIP From
  and/or P-Asserted-Identity headers), the number actually entered the PSTN
  from a data center in New York with the number 212-555-1111.

• Had the calling company presented their billing identifier as 212-555-1111, they
  would have been billed the interstate rate for calling and for this
  example let say that the call would have only cost $3. (And yes, for those of you outside the US, this is true, calls within a state are more expensive than calls between states!)

• Naturally, the calling company wants to present its Caller ID to the
  end user as its own number of 617-555-1111 but it is contracting with the
  SIP telephony hosting provider in part to get cheaper rates. The hosting provider would
  naturally like to be billed at the interstate rate versus the intrastate rate
  when the calls enter the PSTN from their data centers.

There is the disconnect. PSTN billing is still tied to the identity of the originating number as seen through the carrier’s billing systems. Yet with SIP, the caller identity can be different from the identity of the PSTN interconnect point. (And yes, “Caller ID” can be spoofed on the PSTN but there are still underlying billing identifiers in the PSTN signaling.) As long as we continue to have tariffs in North America that have different rates based on geography, we’ll continue to have this problem.

A few years back, the folks over at Sonus Networks came up with an idea to address this disconnect - pass a custom SIP header called “P-Charge-Info” in the SIP INVITE that sets up the call and include in that header the SIP address to which to charge the call. They added this header to some of their equipment and some carriers started using it. So in the case above, the SIP headers might include headers like these:

From: <sip:617555111@sipgw.carrier.com>
P-Charge-Info: <sip:2125551111@proxy.hostingprovider.com>

To the person receiving the call, nothing has changed. The “Caller ID” that is presented on their phone is the same identity they saw before. But inside the cloud, the billing identity has been changed.

[Do note that there’s a double-edged sword here. If the hosting provider and the carrier agree to use P-Charge-Info as the authoritative header for billing purposes, and then in my example above the Massachusetts customer starts making a lot of calls into New York they will wind up being charged the higher intrastate rate based on the P-Charge-Info header. So some analysis of calls needs to be done before agreeing to use this special header.]

Anyway, as one of the largest providers of VoiceXML/CCXML hosting, we are always looking at ways to ensure our customers receive the fairest billing. Late last year after one of our carriers suggested we look at using the P-Charge-Info header, we did so and were quite pleased with the results. Subsequently we have now started using this with multiple carriers.

Being strong proponents of open standards, we noticed that P-Charge-Info was not yet registered on IANA’s list of SIP parameters and so we approach Sonus Networks about going through the RFC 3427 process of registering a custom P-header. They agreed and so Tolga Asveren from Sonus Networks and I submitted an Internet Draft defining the P-Charge-Info header. We received a good bit of feedback/critique and our fourth version (-03) was just posted today.

Why are we going through this registration process, which can be quite lengthy? Several reasons:

• Expanded Usage - We would like to use this with other carriers but until this draft there was no specification for exactly how the header should be used. We would like this to be available as something to point other carriers/service providers to.

• Prevention of Further P-header Proliferation - Right now, if someone is looking for a SIP header to pass billing information, odds are they will look at IANA’s registry to see what headers have been added to SIP. Today the only billing headers are those of the 3GPP defined in RFC 3455 which do differ in intent from what P-Charge-Info does. If they don’t like the RFC 3455 P-headers, they may very well go off and invent their own P-header (anyone can) to deal with passing billing information.

  Our concern is that as we all in the industry continue building this IP interconnect, sooner or later we’re going to want to find ways to exchange billing information and if we have six different headers in production use it’s going to be a real headache. In the ideal world we would have a standard header (not a P-header) for passing billing information and probably we eventually will… but that will take some time to get developed and then deployed. In the meantime, let’s document and register what’s out there now so that people can see the choices they have and not necessarily go right off creating yet another billing information header.

• Prevention of Identity Conflict - It’s very possible that someone going off and creating their own SIP P-header could very well choose to call their header “P-Charge-Info”. One would hope that someone creating a new P-header might use Google to search for the header name they are planning to use (in which case they would discover this existing usage of P-Charge-Info), but someone might not and start using it. It could then happen that some day someone out there would wind up needing to interoperate with two different pieces of equipment that both use a “P-Charge-Info” header, but in different ways. IANA registration is designed to help prevent this.

• It’s The “Right Thing” To Do - If you read section 4.2 of RFC 3427 it states very clearly:
  

  Any P-header
  used outside of a very restricted research or teaching environment
  (such as a student lab on implementing extensions) MUST meet those
  requirements [of section 4.1] and MUST be documented in an RFC and be IANA registered.

  Now obviously no one is enforcing this. There are many P-headers in common usage today that are not documented. No one will slap you down and revoke your right to use SIP if you don’t. But it’s the right thing to do for the reasons outlined above as well as to respect the process defined in RFC 3427.

So where is P-Charge-Info in the RFC 3427 process of IANA registration? Well, Tolga and I now have a fourth revision of our Internet Draft out (it’s the fourth because the first was “-00″) and we would very definitely like comment on the draft. Please do send them in via email (or leave them as comments here). Once the document has gone through more review (perhaps another draft or two) we will then continue the steps in section 4.1 of RFC 3427 and request “expert review”. Ultimately the goal is to have the document published as informational RFC so that it can be registered with IANA.

As I said, comments and feedback are definitely welcome. What do you think of the document? of the problem we are trying to solve?

Technorati Tags:
ietf, sip, standards, voxeo, sonus networks, p-charge-info, voip, telecommunications, telephony, pstn

As we mentioned previously, SIPit22 was held April 14-18, 2008, up at the University of New Hampshire Interoperability Lab. SIPit coordinator Robert Sparks has now posted a SIPit22 Summary which provides a view into the types of products that were being tested.

Now the summary is based on surveys filled out by individual vendors, so it relies on their accuracy (i.e. the information is not independently verified) but it makes for interesting reading. (I was personally struck by the fact that 42% of the SIP implementations indicated they had IPv6 support!) These summaries (older ones available, too) are useful in seeing what vendors are actually deploying and developing versus what we are discussing in the IETF. As noted in Robert’s final paragraph, the issue of NAT traversal for SIP still continues to be one of the most challenging aspects of interoperability.

For our part, attending SIPit 22 was very worthwhile as it gave one of our lead developers a chance to really test the improvements we are making to our SIP stack in an upcoming release of Prophecy. Stay tuned for more on that in the months ahead.

If you do have a product that works with SIP, we do strongly recommend that you do consider attending a SIPit event with your product. We’re a huge believer in open standards… and there’s no better way to test how “standard” your implementation is than to be testing it with a wide range of other vendors!

Technorati Tags:
sip, sip forum, sipit, standards, ietf, voxeo, prophecy

Have you ever wanted to send information between two applications using SIP that didn’t fit into one of the existing “headers” that are using in the SIP communication exchange? Did you look through the various headers in RFC 3261 (and associated RFCs) and find that none fit exactly what you were seeking to send?

If so, what you can use instead is what is called a “private header” or “P-header” for short.

Before we jump into P-headers, let’s first take a moment to switch to the world of email. With email, there are so-called “experimental headers” that are sent in the SMTP exchange that all start with “X-”. Here are some examples:

X-Priority: 	Normal
X-Mailer: 	Apple Mail (2.753)
X-Authentication-Warning: 	blogs.voxeo.com: apache set sender to wordpress@blogs.voxeo.com using -f
X-Spam: 	exempt
X-Mail-From: 	<wordpress@blogs.voxeo.com>
X-Source-Ip: 	[(unknown)]

These “X-” headers are added by various programs to email messages that are sent and share the common characteristic that they are all informational. They do not indicate any action that has to occur. They are not SMTP commands. Each of these X-headers are purely containers of information that is passed along in the SMTP message. On the receiving end, the application accepting the SMTP connection can look at those X-headers and then potentially decide to take some action on the received message based on the contents of those X-headers. If a receiving application doesn’t no what to do with an X-header it receives, it simply ignores the header. With regard to the “informational” nature of X-headers, the idea is that if an application ignored all X-headers, the email message should still be able to be delivered. X-headers add extra information and capabilities, but don’t block the basic existing SMTP.

These “experimental” headers are now widely used by mail programs, by anti-spam software and basically all sorts of other programs that interact with email. They are a way to pass along extra information about the email message. Companies, vendors and individuals can all define and use their own X-headers to try out new services and offerings. They don’t need to check with anyone. They can just go ahead and create a new header to try it out. (Presumably checking first to see that their new name didn’t conflict with existing X-headers on their network.)

Back in the world of SIP, the IETF defined a process in RFC 3427 for extending SIP that was similar to the experimental headers of email. The fundamental difference was that in SIP instead of beginning with an “X-” the headers begin with a “P-”, as in “P-Asserted-Identity”. Otherwise, they are very similar. P-headers in SIP must, like the X-headers of email, be purely informational. They cannot be used to create new commands and must not interfere/block the regular transmission of SIP messages. They merely pass extra information that the receiving application can choose to do something with (or not).

Here are some examples:

P-Asserted-Identity: "Dan York" <sip:dan@company2.com>
P-Charge-Info: <sip:proxy1@company2.com>
P-Charging-Vector: icid-value=1234bc9876e;
                   icid-generated-at=192.0.6.8;
                   orig-ioi=home1.net
P-Visited-Network-ID: "Visited network number 1"
P-I-Made-This-Up: "Pretty silly, isn't it?"

Here is how a couple of these would look in an actual SIP message:

INVITE sip:1234@company2.com SIP/2.0
Via: SIP/2.0/UDP proxy.company2.com:5060;branch=z9hG5bK21ghi7ab34
Via: SIP/2.0/UDP sip.company1.com:5060;branch=z9hG4bKnashds7
To: sip:1234@company2.com
From: sip:dan@company1.com;tag=451248
Call-ID: 324817637683475998ababcc10
CSeq: 1 INVITE
Contact: sip:dan@company1.com
Max-Forwards: 50
P-Asserted-Identity: "Dan York" <sip:dan@company2.com>
P-Charge-Info: <sip:proxy1@company2.com>

The application receiving this SIP message can then choose to read these headers and potentially perform some action based on the contents of the headers. (If you are developing your application on our platform and using CCXML, we’ve provided instructions for how to access those SIP headers.) For instance, the “P-Asserted-Identity” header is very often used as the “Caller ID” that is displayed on the end user’s phone or other SIP device.

So what should you do if you want to start using a P-header to pass information via SIP between two applications?

First, you should head over to the IANA registry of SIP headers to see if someone has already defined a P-header that might do what you are looking to do. For instance, P-Asserted-Identity has been defined in RFC 3325. The 3GPP has defined several P-headers related to IMS in RFC 3455.

Second, if you don’t see any defined P-headers, you might want to search the IETF database of Internet-Drafts to see if there are any P-headers in a proposal stage that might work for you. The process of registering a P-header with IANA (so that people know about it and to prevent naming conflict) is documented in RFC 3427 and first involves the creation of an Internet-Draft document. For instance, I have an Internet-Draft submitted on the P-Charge-Info header that we are seeking to register with IANA.

Third, while in the IETF database you might also just want to search for the general type of information you are seeking to pass between applications. It may be that there is some work underway in an IETF Working Group that would wind up creating standard headers (versus P-headers) to pass the information you are seeking to pass.

Finally, if you are unable to find any current or in-progress P-headers, you can create your own. Keep in mind that it should start with “P-” and the general convention has been to put a dash between any words in the name and to use initial capitals, as in “P-Asserted-Identity”.

For more information about creating P-headers, please do read RFC 3427.

Technorati Tags:
sip, ietf, standards

Did AOL, the historical king of walled gardens, just become the biggest public advocate for SIP and open standards for VOIP?

That’s the point David Beckemeyer makes in his blog post challenging my view of the launch of AOL’s “Open Voice Program” as of questionable value. David counters:

However, it is significant for SIP - we now have a household brand not referring just to “VoIP” but referring to “SIP” - the standards-based protocol for VoIP, something neither Skype nor Vonage has done. SIP means interoperability and that’s a good thing for users and the VoIP ecosystem at large. A brand as big as AOL providing awareness for SIP and promoting interoperability is terrific - nobody else is doing it.

He goes on:

It’s important to note that we have never been able to do this so with Skype, even though the service has been out for four years or more - they could have offered this years ago.

Indeed… point taken, David.

While many of us in the VoIP industry have been trumpeting SIP as the VoIP protocol of choice for years, and work of groups like the SIP Forum has helped in that, outside the walls of our telecommunications/networking community a poll of people would probably find that pretty much no one has heard of “SIP” as it relates to VoIP.

On one level, why should they? It’s just a protocol and at the end of the day it shouldn’t really matter. People just want their phone to work… and SIP and other protocols are just the underlying plumbing.

But for those of us who passionately want to see that plumbing be based on open standards like SIP, David’s right… putting the AOL brand behind SIP is a good thing to help raise awareness of SIP and the need for interoperability. AOL is letting users connect any SIP device to their network. Any device. Not just “certified” ones… not “recommended” ones… any device.

It’s a beautiful thing.

And maybe, just maybe, the people who do start using the service will enjoy the freedom of choice that comes with interoperable open standards and maybe… maybe… they’ll start asking for that interoperability in other services. <rosecoloredglasses>Is that too much to hope for?</rosecoloredglasses>

And so while I still may wonder how many people will really use AOL’s service, as a huge believer in open standards… working for Voxeo, a company who bases its existence on open standards… as a believer in the messy but open process of the IETF to create SIP standards… as an advocate for open protocols and interoperability… as all of that I have to join David in applauding AOL’s move. Thank you, AOL, for jumping behind SIP and helping, in some small way, to help tear down the walls that have locked in telephony for way too long.

Go AOL! Go SIP!

Technorati Tags:
aim, aim call out, aol, aol open voice, sip, ietf, standards, voip, telephony, telecommunications

It’s hard these days to escape news about peer-to-peer (P2P) applications and the concerns raised by service providers about those P2P apps. With recent FCC hearings about Comcast “delaying” P2P traffic, the topic is getting a great amount of discussion across the blogosphere and in the media in general. To look at the issue from a technical point-of-view and see if there are technical solutions that could help in the area, the IETF recently announced a “workshop on P2P Infrastructure” to be held on May 28th on the campus of MIT in the Boston area.

From a voice perspective, any kind of latency / delay is something to be avoided at all costs. As researchers look at P2P SIP as a future deployment model, getting the infrastructure right is a key factor in the future success of P2P SIP. Now, serious deployments of P2P SIP won’t happen for some time now (probably a few years), but the reality is that it will also take some time for any technical solutions to: 1) work their way through IETF; 2) be incorporated into vendor equipment; and 3) actually be deployed in service provider networks. So now is really a good time to get started.

If you are interested in P2P applications in general and are in the Boston area (or can get there), please do consider attending this workshop. There is a wiki page for the workshop that will be updated as the workshop gets closer and a ‘p2pi’ mailing list that is open to anyone to join.

Here is part of the announcement that explains what the workshop will be about:

The Real-time Applications & Infrastructure (RAI) Area Directors, Jon
Peterson and Cullen Jennings, would like to announce an IETF workshop on
P2P Infrastructure to be held on May 28, 2008 at 50 Vassar St, Room
34-101 on the MIT campus in Cambridge, MA USA.

Several large ISPs have encountered issues with P2P traffic. The
transfer of static, delay-tolerant data between nodes on the Internet is
a well-understood problem, but traditional management of fairness at the
transport level has largely been circumvented by applications designed
to achieve the best end-user transfer rates. This results, at peak
times, in networks running near absolute capacity, and in which all
traffic incurs delays; the applications that bear the brunt of this
additional latency are real-time applications like VoIP and Internet
gaming. This has led to need for further discussion of the proper
approaches to P2P application development, and infrastructure management
in environments where P2P is commonly used. This workshop intends to
discover where additional IETF standards work is needed, or existing
work might be reapplied, to alleviate these difficulties. In particular,
the workshop will draw on the experiences of Comcast and BitTorrent,
representatives of both of whom will present their perspectives on the
problem space.

Example solution discussions might include, but are not limited to:
deployment of application servers or caches to reduce network load; new
rendez-vous mechanisms to optimize P2P network topology; enabling
applications to signal their bandwidth needs (and priority or lack
thereof) to networks; enabling networks to signal bandwidth constraints
to elastic and inelastic applications; and, new approaches to fairness
that are coupled with incentives for applications. Contributions from
subject matter experts in the problem and solution space are
welcome. The primary outcome should be a direction for one or more IETF
efforts exploring the best practices for addressing these challenges.

The organizers would like to stress that this is a technical workshop
exploring engineering issues and practices. The public policy
implications of P2P applications are not in the scope of this workshop.

If you would like more information about how to participate in the workshop, please read the full announcement.

Technorati Tags:
ietf, standards, p2p, peer-to-peer, p2psip, sip, voip, comcast, bittorrent, Internet, networking

This video is about six months old, but after a friend reminded me of it, I realized that I had to blog about it. I mean, how can you not be amused by “The Day The Routers Died…” (once you adjust to the use of the European “rooter” vs the American “rowter”):

Note that if you click on the video and go to the YouTube page, you can click on the “More Info” link and get the full words.

For those not familiar with RIPE, it is an open organization for Europeans interested in the Internet and through the RIPE Network Coordination Centre provides allocation of IP addresses, domain names and other aspects of Internet “policy” (such that there is) in the European region. This song was performed at their last meeting (RIPE 55) and their next meeting, RIPE 56, is coming up May 5-9 in Berlin.

Kudos to whoever spent the time to come up with these lyrics and to the gent performing it! Nicely done.

Technorati Tags:
ipv6, videos, ripe

What 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…

Technorati Tags:
sip, sip forum, sipit, standards, ietf, voxeo, prophecy

In 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!

Technorati Tags:
IETF, sip, standards, aprilfools

If you want to get a sense of what an IETF meeting is all about, I’ve now uploaded a set of photos from the recent IETF 71 meeting in Philadelphia.

If you get the sense that we spent a lot of time sitting in chairs… you would be correct! It’s all about chairs, hallway conversations and, of course, (for those of us with no aversion to speaking publicly) the microphone! (Unless, that is, someone relocates the mic.

IETF meetings also naturally involve a whole lot of email, even while we are there at the meeting. Given that we had a very solid wireless network (even if access points were sometimes on chairs) and we also had a 100 Gbps (yes, you read that correctly - 100 Gigabits per second!) connection to the Internet via Comcast you’ll notice that basically every picture has a very high density of laptops. If there were 1300 people at the meeting, probably 1295 of the attendees had a laptop there with them.

Technorati Tags:
ietf, ietf71, philadelphia, comcast, standards

Want 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.

Technorati Tags:
sip, p2p, p2psip, sipeerior, ietf, sipdht