Posts categorized "SIP"

ietflogo.jpgAs I wrote over on Voxeo's "Speaking of Standards" weblog, one 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!

Jonathan Rosenberg has now offered a definition in a new Internet-Draft titled "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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The EComm 2008 Interview with Skype's Jonathan Christensen should be required reading...

42F19C6B-67C5-433E-91B4-641B9323CD48.jpgAs we enter into the final month before eComm 2008, I would suggest that the interview with Jonathan Christensen, Skype's general manager of audio and video, should be required reading for anyone seriously interested in this space. Why? Well, in part because Jonathan Christensen does provide some good information about what Skype has done and is doing but also because it provides some good insight into what one of the people driving Skype's agenda is thinking about this space. Take one of the final paragraphs where he answered Lee Dryburgh's question about what he saw as the the future of communications (bold emphasis added by me):
Well, a big question I guess and, having worked on the space for quite a while, I think that it's only going to get more interesting over the coming years since, well, like this open spectrum for example. You know, I just have to reiterate, I think that anybody who has not figured out that the Internet is the platform and that there isn't any such thing as walled gardens that will survive, or sub-networks [such as AOL tried] that are going to survive, those people are doomed. The intersection of these worlds is going to be chaotic. It's going to be violent. It's going to be messy for a while but it is going to happen, and the Internet will survive as the one open platform. You are going to see a trend towards extreme innovation at the edges - on the devices, in the PC platform, in software, all around the edge of the Internet.

I think that you are only going to see further disruption of the telecom industry and the emergence of totally new businesses that we can't imagine today. I think that [the] net result, that drives me every day, is that we're going to have this very rich, open, cheap and accessible communications. This is going to be not just a game changer for the telecom industry, but will be a change agent for all of humanity. So, a platform that allows us all to see each other and hear each other more clearly maybe makes us a little bit less crazy, less polarized and more open as a world society.

Good stuff... and the whole interview is worth a read. Given my recent criticism of Skype, I'm particularly pleased to read the comments I emphasized in bold. Jonathan Christensen will be giving one of the keynotes at eComm 2008, March 12-14 in Silicon Valley and if you haven't considered going, I would encourage you to do so. It should be a great event!

P.S. I also wrote about this interview in relation to SIP over on Voxeo's "Speaking of Standards" blog.

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I'll be speaking at Ingate's SIP Trunking Seminars at IT Expo in Miami next week

button_Miami08.gifIf any of you will be in Miami next week for Internet Telephony Expo, I will be speaking on VOIPSA's behalf at Ingate's SIP Trunking Seminar Series held in conjunction with IT Expo. Predictably, my session from 8:30-9:45am on Thursday, January 24th is titled "Seminar/myth 1: VoIP is not secure".

If you are going to be down at IT Expo, do check out the full schedule for Ingate's SIP Trunking Seminar Series. They have a good range of speakers and the seminars are free.

If any of you are attending either IT Expo or the SIP Trunking Seminar Series, please do drop a note as I'm always interested in meeting readers.

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A SIP phone for the iPod Touch! (Just add microphone...

Fascinating development on the Apple frontier... in late December some developers posted information about a SIP phone for the iPod Touch! They included this helpful demonstration video:

The team has obviously received a lot of questions and has therefore released a lengthy FAQ list. If you have an iPod Touch, you can download the software. Of course, you really need a microphone to use it... which the Touchmods folks are building.

All in all an interesting development. I look forward to seeing how it moves along!

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SIP phone soon to be available on iPhone and iPod Touch?

2B23DB8E-7E95-4B93-A7CB-A55877BD20BA.jpgWill there soon be a native SIP client on the iPhone and iPod Touch? Dameon Welch-Abernathy writes on his VoIP weblog that some developers have gotten a basic SIP stack working on the iPhone and iPod Touch. The limited details available are over on The Unofficial Apple Weblog:
iPhone hacker eok writes to let me know that he and Samuel have gotten SIP registration and signalization working. They took a few mobile terminal shots, but the real work is being done via ssh. Samuel is working on connecting the audio in/out to the pjSIP. If you have iPhone or iPod touch coding skills and want to get involved in the project, connect to #touchmods on irc.undernet.org. It looks like most of the work will be done on European time.
As you can see in the screenshots, this is still very early in the development. Still, it's great to see this kind of development taking place.

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Introducing "Speaking of Standards", a new Voxeo blog about industry standards, IETF, W3C, SIP Forum, etc.

200711292028A large part of why I have NOT been writing here all that much in the past few weeks is that I've been busy in my new role with Voxeo working on a corporate blog portal. I've been covering a bit of that odyssey over on my Disruptive Conversations blog as well as in my weekly reports into the For Immediate Release podcast. It's been a great amount of work but also a lot of fun - I've been very lucky to have a colleague who does amazing things with CSS and graphics, and so the sites look a whole lot better than they would if I were left to my own devices.

I'm very pleased to say, now, that we've reached the point where I'm willing to link to our work and talk a bit about what we are doing. The main blog portal is the predictable "blogs.voxeo.com" but the weblog that we're really starting to use and could be of interest to readers of this blog is our "Speaking of Standards" blog found at:

http://blogs.voxeo.com/speakingofstandards/

I've obviously been very occasionally writing here about standards and some of that may continue, but I expect most of my writing on the subject will now occur over on this new Voxeo weblog - and I'll naturally be writing more on the subject. We'll be writing about the IETF and SIP standards, but also the W3C and standards such as VoiceXML and CCXML that I've never covered at all here. We'll be linking to events and tutorials we find and generally providing whatever information we can about standards affecting our industry, as well as Voxeo's views and implementations of those standards.

Why would Voxeo sponsor a weblog about standards? Primarily because the company and our products are all about open standards - which was one of the things that attracted me to the company after they first approached me. I've since learned that they've been leading the IVR industry in adopting open standards. As the products page says in the "Fast Facts" sidebar:

  • 100% Standards based IVR
  • Supports W3C VoiceXML 2.0
  • Supports W3C CCXML 1.0
  • Supports W3C SRGS 1.0
  • Supports W3C SSML 1.0
  • Supports CallXML 3.0
  • First platform with XML call control
  • First platform with XML conferencing
  • First shipping CCXML implementation
  • First SIP/VOIP IVR platform

Not bad, eh? Add to that the fact that our CTO (my manager), RJ Auburn, chairs the W3C's Working Group on CCXML and we've hired other folks involved with standards efforts... all of that is why we added a weblog on standards.

So if you would like to see our view on industry standards, find tutorials about various standards or learn about standards-related events we may be attending, I would invite you to come on over and check out "Speaking with Standards" - or subscribe to the RSS feed. While I (and others) will still be working on improving the site, it's mostly done and I'm delighted to be able to return to writing more. Let us know what you think!

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New Facebook voice app: VoxCall lets you do free calls between SIP phones/numbers

200711200958By way of my Facebook NewsFeed this morning, I learned that several friends had installed a new Facebook app called "VoxCall" (must be logged into the walled garden of Facebook to see the link). A quick Technorati search brought me to Alex Saunders' blog post on the subject which clued me in to the fact that this was from the folks at Voxalot, some of whom I'd met down at Fall VON in Boston.

[Side Rant: This also shows the inherent weakness and stupidity of Facebook's current implementation of "groups". The Voxalot folks had posted info about this app in both the news and wall of their Facebook group, but of course I would never see it unless I just randomly happened to go there. Had they sent a message to all group users, I would have seen it in my Facebook Inbox, but it would be nice if instead Facebook had some way to notify you that you had new info in the groups to which you subscribe.]

The VoxCall app is basically a "click-to-call" app that makes use of Facebook's directory. You simply click on the name of someone else who has the app installed and, like many click-to-call apps, you are called first and then the other party is called and the connection is made.

An interesting aspect is that VoxCall works with SIP URIs (addresses). When you install the app you have to enter your SIP URI at which point you then receive a call on that URI where you are asked to enter the PIN displayed on the screen:
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It's actually a pretty nice way of authenticating the endpoint. Given that Voxeo's a VoIP application platform company, we naturally all have SIP URIs for our extensions (sip:[email protected] for me) so it was easy for me to sign up. Users of Gizmo would likewise have a SIP address, as would users of many other VoIP services. If you don't have a SIP URI, Voxalot has a suggested path to get one on their VoxCall FAQ. (One thing I don't completely understand is why you would need to do their step #2, Register for a VoxPremium account, if you already get a SIP URI from the Voice Service Provider you signed up with in step #1. But maybe the point is that some of those VSPs won't give you SIP URIs... ?)

Once registered, the process is quite simple. You have a "Call Friends" tab that is shown below (complete with some advertisement being blocked by the local proxy server that I run that blocks ads from typical ad-serving sites):
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You simply click on the person's icon and the call process starts. First it calls you, then it calls the other party. No charges incurred by anyone outside of whatever inbound connection fees we would normally pay (in my case, none). I called Alec and so my page changed to show his picture and the fact that I was calling him:
200711201005
Alec and I had a good chat with surprisingly good audio quality given the convoluted path our call was taking. I was on a Polycom IP phone connected across the Internet to Voxeo's SIP servers in Florida. The call went across some network cloud to Alec's TruPhone number (which has a SIP URI) which wound up ringing his mobile as he was driving along the 401 somewhere in southern Canada. Audio quality was quite good and didn't seem to have any real issues in the 5 or 10 minutes we chatted.

The VoxCall app also has an Echo Test number you can call to hear the latency and has some conference rooms that I have not yet tried.

Overall, it's an interesting app, although I guess my basic question is simply this: will I use it? As I wrote earlier, the phone is no longer as critical of a communication tool for many people, myself included. When I think of Facebook, I think of it as a place for email-ish communication. If I need to reach someone urgently, I have used Facebook as a place to get a phone number from in the past. Will I think to use to it place a call in the future? I don't know.

There are a couple of barriers to that, really. First, the app only works with people who have it installed. Second, to install it you need a SIP URI and the whole concept of SIP addresses is only really now starting to come to people's attention (outside the early adopter crowd). Third, initiating the call requires going into the VoxCall application page inside Facebook to click on the person's icon to call. It would be nice if it could be done simply from the list of friends that you have. (Having said that, it's actually easier to simply go into the app page than it is to search through Facebook's friend list and then go into their profile to then click on a link below their picture.)

The nice thing about the app, though is that it does use the Facebook directory. As Alec puts it:

Perhaps the biggest differentiator for Voxcall is simply that it hooks into a directory that a lot of people know and use.

As Facebook continues its climb in popularity and moves onward toward the goal of being your definitive "portal" to the Internet, this VoxCall app (and others like Alec's own Free Conference Call app) help connect in voice to the communications mix (for those who still want/need to use it).

In any event, kudos to Voxalot to bringing out another voice app on top of Facebook. It's good to see the platform being used for voice. As a advocate for SIP and open standards, I applaud apps that promote the use of all things SIP. Give it a try. What do you think of it? (Feel free to give me a call if you are a Facebook friend of mine.)

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I'll be out in Vancouver Dec 2-7 for the 70th meeting of the IETF.

200711191406Just confirmed travel plans today - I will be heading out to the 70th meeting of the Internet Engineering Task Force (IETF) in Vancouver, British Columbia, Canada, from December 2-7. If any readers will be out there (either for the IETF or in Vancouver in general), please do drop a note and let me know. This will be my first meeting in my new role with Voxeo and I'm very much looking forward to renewing old acquaintances and also getting more directly involved with the work of the IETF.

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Did you know RFC 4733 had replaced/obsoleted RFC 2833 for DTMF signaling in SIP?

Did you know that RFC 4733 replaced/obsoleted RFC 2833? I just learned this myself through a SIP Forum mailing list exchange the other day. For those not aware, RFC 2833 and now 4733 define methods of carrying DTMF signals (and other similar signaling) in RTP streams separate from the main audio component of the RTP stream. A typical example of use might be where you were using a highly-compressed audio codec for audio between two SIP endpoints where the high degree of compression might make it challenging for the DTMF tones to be correctly interpreted on the receiving end. Using "RFC 2833 compliant" signaling, the sending SIP endpoint would send those DTMF tones as separate packets within the RTP stream.

My key takeaway from learning about RFC 4733 is that we should really be talking about "RFC 4733 compliant" signaling... but given that the industry is really only now starting to really talk about "RFC 2822 compliant" signaling, I'm not sure I expect to see that happening anytime soon.

Anyway, here's the abstract from RFC 4733 - you can naturally read the rest of the document to understand more:

This memo describes how to carry dual-tone multifrequency (DTMF) signalling, other tone signals, and telephony events in RTP packets. It obsoletes RFC 2833.

This memo captures and expands upon the basic framework defined in RFC 2833, but retains only the most basic event codes. It sets up an IANA registry to which other event code assignments may be added. Companion documents add event codes to this registry relating to modem, fax, text telephony, and channel-associated signalling events. The remainder of the event codes defined in RFC 2833 are conditionally reserved in case other documents revive their use.

This document provides a number of clarifications to the original document. However, it specifically differs from RFC 2833 by removing the requirement that all compliant implementations support the DTMF events. Instead, compliant implementations taking part in out-of-band negotiations of media stream content indicate what events they support. This memo adds three new procedures to the RFC 2833 framework: subdivision of long events into segments, reporting of multiple events in a single packet, and the concept and reporting of state events.

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Skype and secure SIP? (Why would I see this message?)

200710261520Whenever I'm using Skype, I have the "Display technical call info" setting enabled so that I see technical stats about the calls I am on. Those windows tend to stay around after a call... and I noticed this one still around with an identity of "securesip". (click on the image for a larger version) I've tried to replicate this with calls that I've recently made to see if I could get the window again, but can't seem to do so. Anyone know why I might be seeing this?

I'm curious...

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