Follow the contest on Google Plus
When Gerd Graumann last visited with us in December, we talked about Speech Technology Group‘s  natural text-to-speech voices and launched the speech humor contest. In this session he will announce the winner and play the funniest submissions.

The discussion will be on speech recognition technology, which has lately taken center stage though the widespread mobile adoption. Gerd will share with us how Speech Technology Group has taken a powerful 64-bit ASR that is used by thousands of companies exclusively within the unified communications space to work now with every platform that supports the industry standard MRCP interface.

Kevin Fleming and Malcolm Davenport call in from Europe for a session on the new Digium phones announced a few days ago.

More about the phones from Dave Michels on No Jitter

The SIP school offers training and certification in SIP and other VoIP technologies. Use VUC discount code VUC10 for 10% off!

As the provider of the global standard in SIP training and certification, The SIP School has taught thousands of employees in the telecommunications industry how to better  support their clients, products, and services. Until recently, students training to become an SIP School  Certified Associate (SSCA®) were instructed in their first session to create a SIP address with any free service. Today, The SIP School announces another option by working with OnSIP as their SIP  service provider – leveraging the OnSIP API to provision each student with a SIP address on thesipschool.com domain.

This is the original press release.

Joining us this Friday is Vidtel, a cloud-based video conferencing service that is easy to use and price-disruptive based upon Any to Any model. Spanning boardrooms, mobile workers and everything in between, the Vidtel Meet Me service allows room-based video conferencing systems, executive desktop video systems, PCs/Macs, smartphones and tablets to work together. Any combination of SIP, H.323, Skype and Google Talk technologies can interoperate on demand with Vidtel’s service. The cloud service is targeted to the small-medium enterprise market, priced for any-sized budget and is as easy to use as a telephone.

Scott Wharton, Founder and CEO of Vidtel, is a recognized leader in the VoIP and video industry and will join the call to discuss video landscaping and why any-to-any video calling will be important in the coming years.

From 1999 to early 2008, he was the VP of Marketing at BroadSoft, Inc., the leading developer and marketer of VoIP application servers, which also handle video IP calling. Scott made significant and enduring contributions leading to: the dominance of the BroadSoft brand in the voice application space, the company’s acquisition of 9 of the world’s 10 largest telecom service providers as customers, and the expansion of the company’s presence in over 60 countries. Scott also defined a new product category for the telecom industry, a next- generation applications layer, establishing BroadSoft’s global leadership in the VoIP applications market.
In 2003, Scott was honored with the Frost & Sullivan IP Telephony Executive of the Year Award. It is presented each year to the individual who has excelled in developing and executing a business strategy allowing his or her company to gain market leadership. In 2006, Scott was recognized as a “Top 100 Voice” in the IP Communications Industry by Internet Telephony magazine.
Prior to BroadSoft, Scott was Vice President of Marketing at VocalTec Communications, the pioneer and a leading developer of VoIP software solutions. From 1995 to 1999, Scott spearheaded numerous groundbreaking initiatives, including the development of the first commercial VoIP product and the creation of the VoIP service provider industry. In the 1990s, Scott created one of the first IP video products for the consumer market as a senior marketing exec with VocalTec, the pioneering VoIP company. Scott holds an MBA from Yale University.

We will be giving away 1 OBi110 and 1 OBi100 to “lucky” listeners. See posts on Google Plus.

If you don’t win one, you can order an OBi100 VoIP Telephone Adapter and Voice Service Bridge from Amazon, which will also throw a buck or two into the VUC beer fund.

Obihai is just getting going, but are already enjoying the popularity the OBi has received from end-users, VARs and VoIP SPs. We will talk about what they’ve done in the SIP, XMPP, P2P and social networking technology space and how this is related to the explosion in social networking apps, the popularity of VoIP services and the ‘un’ popularity of the land line service.

They sent me an OBi110 evaluation unit. I connected and plugged in the lightweight, white plastic box, called the echo test number, and it worked perfectly, with a low lag time from here in Southern California. Fine, but for us logtime VoIP users, not exactly a revelation. However, the ease of connecting to a Google Voice number suddenly makes this very reasonably-priced ATI very interesting.

Sipgate CEO Thilo Salmon returns to VUC this week with Diana Cionoiu, to tell us of their use of Yate. We see on the Yate site that extensive use has been made of Sangoma hardware products.

Press release:

The open-source telephony engine Yate is making inroads into what used to be the domain of proprietary equipment provided by large telecom suppliers. Following the certification of Yate SS7 stack by Deutsche Telekom’s test lab, Sipgate has started to deploy Yate at the core of it’s softswitch. The decision came following extensive tests to determine stability, scalability as well as resilience.

“The open source telephony platform was easily integrated in the infrastructure of our softswitch”, said Thilo Salmon, CEO of Sipgate. “We are mostly impressed by the quality and scalability of Yate technology.”

“We are pleased that sipgate has chosen Yate for implementing their telephony softswitch”, said Diana Cionoiu, managing director of NullTeam. “This is a solid recognition of our 7-years development of Yate platform”

Part 2
After Hours 326: Death of the Desk Phone by VoIP Users Conference

In hour #1 of a double-header this week we will be joined by Khris Kendrick, Senior Director of Business Development at Grandstream Networks. Many of you probably have experience with Grandstream’s well-established range of SIP hard phones and ATAs.

While Grandstream is long known as a market leader in affordable SIP end-points, they have stepped into new areas of business including video capable phones and IP surveillance cameras.

VUC regulars Michael Graves and Karl Fife have recently been using their latest video desk phone, the GXV3175. A very interesting desk phone with a 7″ touch-screen LCD display, the GXV3175 offers a diverse range of multi-media capabilities. Additionally, VUC sponsor OnSIP recently published a review of the GXV3175.

This is just one of the topics that will surely be discussed. Bring your breakfast, lunch or dinner & bring your questions.

By the way, Grandstream is running a contest to give away a GXV3140 phone and GXV3615 IP Cube Cam. Just drop by their Facebook page and like it to enter.

Some related links:

• Grandstream IP Multimedia Phone Channel on VoIP/Telecom Industry Leading publisher of Internet Telephony TMC
• Twitter: http://twitter.com/#!/GrandstreamNet
• Facebook: facebook.com/GrandstreamNetworksInc
• Website: www.grandstream.com
• Onsip Review of GXV3175
• Onsip review GXP2100
• CRNs 11 Key Videoconference/UC Solutions

We welcome John Riordan, CTO, and Charlotte Oliver, Systems Engineer, from Junction Networks for this talk. IPv6 has serious benefits for SIP, so Junction Networks has made it a priority to begin making their network and systems IPv6 ready. Currently, they are directly connected to the IPv6 Internet via multiple providers and have established network infrastructure and redundancy. Also on the topic of OnSIP platform preparations for the future, Junction Networks will discuss their patent-pending process for scaling SIP proxies., including their recent milestones in establishing an n+1 architecture, and how this supports their end goal to deliver high quality service to OnSIP customers.

On this session, Pascal Doré of Media5 Corp joins us to talk about the evolution of their products since his last visit over one year ago. Media5 Corp’s products include SIP clients for IOS and coming soon, for Android.

Dave Michels (@DaveMichels on Twitter) will finally clue us in on Lync, a topic he’s covered many times in his excellent blog that you can follow at www.pindropsoup.com.

Microsoft’s VoIP solution known as Lync is changing the conversation at the enterprise. Rarely does a single product cause so much controversy and impact. Dave Michels will provide an overview of Lync, and an analyst’s view of why its stirring the pot.

Mike Oeth of Junction Networks called in to tell us about their newest development, “Get On SIP”. We also spoke of the recent EC2 outage and some ITSP-related issues such as proxies and SIP registration.

getonSIP is a fast, easy and FREE way to register a SIP account, allowing you to communicate using audio and video over the Internet using the SIP protocol. getonSIP uses the same technology as Junction Networks’  hosted enterprise phone system, OnSIP, so you get plenty of great features, including HD audio, voicemail to email, and much more.

http://www.getonsip.com

Chris Matthieu on Tropo.com, great added comments by Tim Panton. Michael Graves, just back from NAB with a few notes. Karl Fife on WebOS (just kidding!)

Bad Wifi at the trade show, new Siemens Gigaset handsets and more DECT comments. Profiles on mobile devices, sharing books on Kindle, Nook, Android, iOS, etc.

This month, many exclusive news items have been obtained by the VUC and we will be sharing these scoops with you this Friday.
photo: Scott Beale / Laughing Squid

Dan York (insert typical remark about not him needing any introduction) jumped on the news that Google Voice was offering up SIP URI calling on March 7th, noting as his source a tweet from Aswath Rao (crediting @truvoip), all familiar contributors to the VUC. Then, Dan and many of us found it wasn’t working and he posted “Did Google Hang up…?” and then just two days later: “Google Voice Via SIP – It’s Dead, Jim”.

Plenty to talk about, but there’s a lot more news in Dan’s blog. Can he confirm that the AT&T/T-Mobile acquisition will make a new brand name in telephony: “T&A”?

Tune in and see!

OnSIP + Highrise

For those who adore SIP, open standards and open source, join us and meet AG Projects team, the makers of Blink and hear about the project, its inception and its future. Our guests are Adrian Georgescu, Dan Pascu and Saul Ibarra. These folks have done great things in the past, and we’re anxious to hear more about their latest work.

Adrian Georgescu, AG Projects founder and CEO, calls Blink “the new kid on the block” among SIP clients. Launched just one year ago, it managed to become for many in a short period of time their favorite SIP client on the Mac and now effort is underway to provide the same cool experience on Windows and Linux. Blink differentiates itself from its competitors by a generous feature set: g722 wide-band Audio from day one, File Transfer, Instant Messaging and Desktop Sharing, all those cool things SIP was all about, yet everyone stopped at VoIP. The lack of dial-pad also stirred some waters, it was the first soft phone to say goodbye to the classic dial-pad, a telephony oriented interface and offer instead a productivity oriented contacts driven interface. AG Projects has launched this month SylkServer, an open source product that complements all Blink features on the server side, and the two of them can provide now a Skype like multi-party conference experience by using pure SIP. With an impressive roadmap, proven capabilities and real customers behind, can AG Projects now make a difference in the crowded space of SIP and VoIP?

SylkServer allows creation and delivery of rich multimedia applications accessed by SIP User Agents. The server supports SIP signaling over TLS, TCP and UDP transports, RTP and MSRP media planes, has built in capabilities for creating ad-hoc SIP multimedia conferences with HD Audio, IM and File Transfer and can be easily extended with other applications by using Python language.

iNum stands for international Number and is an initiative managed by Voxbone to develop an area code for IP Communications within the +883 global country code created by the ITU (International Telecommunication Union) started in 2008.  This will be an update to the earlier visit to the VUC made by iNum team members in January 2009.  Hugh Goldstein, Director of iNum Business Development will share current information and plans for 2011 and how VoIP users can leverage and support the iNum initiative in their own businesses and as individuals. Note that iNum supports voice, SMS and advanced multi-media services

Here’s Voxbone’s list of carriers who support iNum. (We notice that Orange Switzerland is not on the list, but we know that they can route iNum calls. They do this as surcharged numbers.)

Congratulations to wolrah, our winner!

As we approach the US Thanksgiving holiday we are given pause to reflect upon our many blessings. While many of us exist in a wideband realm where clarity of voice leads to universal understanding, there remain a great many who suffer through the narrowband reality of the past. These poor folk suffer not only a startling lack of clarity, they are typically tethered, bound to their toil by wires.

Gigaset Communications aims to give a hand to some poor unfortunate soul, bestowing upon them the gift of clarity*. This gift takes the form of a Gigaset S675IP SIP/DECT starter system. To further aid in the mobility of the downtrodden they also offer the S79H expansion handset, enhancing the basic set.

These tools, HDSP capable…or HDVoice capable for the followers of Plcm….promise to uplift up their users with both clarity of voice and mobility of action. Such outstanding clarity that they might understand The Words.

However, such clarity will not be bestowed upon the idle or laggard. It will be granted only to the one who proves themselves most worthy at the appointed time. A question will be asked. These gifts will be given to one of those who answer the question correctly.

Look to the mailing list. Watch for the signs. Then be on the call Friday, November 26th.

*For regulatory reasons this offer is restricted to North American listeners only.

SIP Communicator – the Java VoIP and Instant Messaging client.

SIP Communicator is an audio/video Internet phone and instant messenger that supports some of the most popular VoIP and instant messaging protocols such as SIP, Jabber, AIM/ICQ, MSN, Yahoo! Messenger, Bonjour,  and a whole lot of other useful features. In the latest builds, SIP Communicator has g722 capability, making it a choice for use with VUC via ZipDX.

SIP Communicator is completely Open Source / Free Software, and is freely available under the terms of the GNU Lesser General Public License.

SIP Communicator has binaries built for Mac OS X, Windows and several linux distros, and the source is available on the SIP Communicator downloads page.

Double innovation, two topics this week as Counterpath has two big stories. Todd Carothers, VP Product Management and Mike Doyle, VP Technology will be on board to tell us more about these two developments and to answer your questions.

Counterpath was granted a patent (# 7,809,381,) for presence detection in mobile and fixed broadband networks.

By providing applications with a graceful way to automatically select the best network connection based on the user’s location, the patented mechanism enables services such as least-cost routing and minimizes dropped calls by streamlining the process of transferring calls from the cellular macro network to Wi-Fi or a femtocell.

The new patent announced today leverages these two patents to enable the delivery of the voice and data services based on the end user’s location. For example, when a person moves from a mobile network into an IP-based network – such as one in a home or office – the combination of these three patented mechanisms allows the home or office phone to automatically assume the user’s mobile identity, or mobile phone ID. When the user leaves the home or office environment, the phone is seamlessly disconnected from the mobile identity and moved back to the mobile network.

Counterpath Bria for Android

Compatible with any smartphone running Android V2.1 or later, Bria Android Edition’s features include:

    -   Support on any SIP-compliant server to enable rapid implementation
    -   Multi-tasking support for background operation, such as fielding
        incoming calls while using other applications
    -   Automatic codec selection to ensure optimal call quality
    -   Deskphone-class options such as rejecting, holding, forwarding,
        merging and splitting calls, as well as attended and unattended
        transfers
    -   A detailed call history pane that displays dialed, answered and
        missed calls, along with ability to delete entries
    -   Advanced security settings, including audio encryption
    -   NAT traversal to balance security and ease-of-use
    -   The ability to work with the native dialer
    -   Integration of user contacts stored on the device
    -   Voicemail
    -   Optional customized branding for graphic assets and SIP settings
        available for enterprises and telephony providers.

Many VUC members are testing the beta now and they will be able to interface with the Counterpath folks during this call.

Actually, it was before hours, but who’s counting? Talk about Apple TV, Cisco wanting to buy Skype, Skype for Business (was SIP for Skype), bluetooth headset woes and much more.

SIP for Skype is now called “Skype Connect”. We’re wondering about the cost of the service. At one point I heard there would be a base charge for Skype Manager and then a charge for each user, plus a per channel charge. If this is the case, I can only say “Get a Life, Skype!”

Some intense examination of the requirement to tell people you are recording a call with the regulars while waiting for FS bridge to successfully UNmute it self. Note that ZipDX implemented the warning when you connect to the bridge if recording is on. Some reminders about *6 to toggle mute state.

The origin of “Look to your skies for a warning! Keep watching the skies!” Naw, I cut that part but I wanted to add the link in this post, because I didn’t get more info from FS about what to put here. Great movie, “The Thing from Another World”, featuring James Arness (“Gunsmoke”) as “The Thing”. One of the best science fiction pictures from the fifties, and one that helped define the genre.

From yet another world, (the Freeswitch bridge), Pacman sounds, and then belched forth the actual meat-space aural personna of the like of Brian West, Michael Collins, Anthony Minessale, Darren Schreiber and a few less-stellar voices…

From their site at Freeswitch.org: “Created in 2006 to fill the void left by proprietary commercial solutions. FreeSWITCH also provides a stable telephony platform on which many telephony applications can be developed using a wide range of free tools.

FreeSWITCH was originally designed and implemented by Anthony Minessale with the help of Brian West and Michael Jerris. All 3 are former developers of the popular Asterisk open source PBX.  The project was initiated to focus on several design goals including modularity, cross-platform support, scalability and stability. Today, many more developers and users contribute to the project on a daily basis.”

This coming Friday, August 13th will be yet another Voip Users Conference double-header. Starting at our usual 12 Noon EDT we have an overview of the Freetalk Connect SMB PBX featuring Skype integration.

Then immediately following, at 1pm EDT, we have Chris Veazey, VP Engineering of Blink Mind, to discuss the current state of the industry with regard to multi-media phones and SIP-based video calling.

Blink Mind has agreed to provide a video conference bridge (MCU) as part of their appearance. Further, as Blink Mind are a Polycom partner, Polycom has graciously provided a pair of Polycom VVX-1500 Business Media Phones on loan. One VVX is at Randy’s location in California, and the other in my home office.

As we aim to explore the capabilities of the VVX-1500 and similar desk phones, the video portion of the Blink Mind call with be limited to CIF (352 x 240) resolution using H.264 compression.

The Blink Mind bridge will allow compatible connections from a variety of clients. If you have access to a Polycom VVX-1500 or Grandstream GXV series phone you will be able to join the video bridge. Also, many larger video conference systems will drop down to CIF resolution, so they too can join the bridge.

If you don’t have access to such hardware you may be able to use a compatible soft phone. Rudimentary testing with the Mirial soft phone last Friday proved successful. Mirial is available for Windows and OSX. Mirial is a commercial product, but a 30 day trail download is available.

Since we have a relatively low-resolution video stream this time we’ll be issuing two kinds of connect details for the Blink Mind bridge. Yes, there will be two different SIP URIs for access to the Blink Mind bridge:

• Some will be able to join in a fashion that allows them to participate in the video stream.
• Others will join the bridge using a non-participatory SIP URI that gives them a view of the video stream but does not add their video to the MCU feed.

In this manner we will avoid having a 16+ way split where everyone looks like a tiny animated smudge on-screen.

If you have a suitable hardware based means of joining the video conference please let me know. While I hate to be arbitrary, there are a limited number of participatory seats available and preference will be given to those who have suitable hardware.

The details of the non-participatory connection and the web stream will be announced to the VUC mailing list early in the week before the call. We’ll also announce a SIP URI for anyone who needs to test that their end-point is capable of joining the Blink Mind video bridge.

Of course, we will join the ZipDX wideband audio bridge to the Blink Mind video bridge. A recording of the video stream will also be available after the fact, along with the regular podcast.

If you don’t have a webcam or suitable soft phone you can still view the video stream. Blink Mind will be providing a Flash-based streaming feed of the call. Since the stream is post-processed it’s a few second delayed from the bridge, but it includes the audio and video. This is a “view only” feed so you will not be able to participate in the call if this is your only means of connection.

If you are joined to the web stream and the ZipDX bridge we’d asked you to be very careful. The audio will be out of sync between these two bridges and it could become very confused for everyone if you allow the delayed audio into the ZipDX conference.

Of course, we will join the ZipDX wideband audio bridge to the Blink Mind video bridge. A recording of the video stream will also be available after the fact, along with the regular podcast

Watch the VUC mailing list over the coming few days. As more details get locked down things will be announced there.

Freetalk Connect wants to put Skype on every phone in your office.
Designed for offices with between two and 50 users, the FREETALK Connect offers  unified communications functionality, including Find Me, Follow Me; a unified voice mailbox; automated attendant and auto call distribution.

Intelligent routing capabilities:  incoming Skype calls, as well as calls over SIP, the PSTN and IAX2, can be routed by the FREETALK Connect to any local or remote Skype user, SIP endpoint, analog or mobile phone.

Set up is simplified, enabling small business users that are not tech savvy to use it quickly and simply, without formal training. Supported telephones are plugged into the company’s network and the device auto-detects and configures them. An on-screen wizard then guides the customer through a few business-related questions that help configure the company’s communication system and enables Skype calling from every supported desktop phone in the office. Adding users and administering the system after installation is just as easy.

Both SIPVicious and Adhearsion have announcements to make. Sandro joins us to explain “svcrash”, a new addition to the tool kit to stop attacks when they happen:

A new tool has been added to SIPVicious – svcrash.py. As the name implies, it crashes something – svwar.py and svcrack.py. This tool is meant to be used by system administrators and organizations that are receiving unauthorized scans on their exposed IP PBX.

Ben joins us to give some details about the evolution of Adhearsion which has had a recent version release:

This release marks the continuation of the platform with some major enhancements and bugfixes:

• Support for Asterisk 1.6 (continued backward compatibility with Asterisk 1.4)
• Added support for ActiveLDAP
• ActiveRecord now works properly with Adhearsion components
• Daemonizing of Adhearsion no longer truncates log files
• Escape commands sent to Asterisk via AGI
• Asterisk Manager Interface (AMI) events now work when daemonized
• Various enhancements for improved Asterisk 1.6 support
• Various bugfixes

John Todd gives a preview of Astricon 2010, we chat about AMOOCON in Rostock next week and about various VoIP trials and tribulations. We have a special VUC discount code for you. Enter code AC10VUC for a significant discount.

We’ve been testing Skype for SIP open beta today. This allows you to connect Skype to your commercial PBX, much as Skype for Asterisk allows connection between Skype and Asterisk. Listen to the recording to learn more about the experiences to  date.

The Incredible PBX: Free Skype, free worldwide SIP, free U.S./Canada PSTN plus rock-solid Asterisk security using the new Zero Internet Footprint™ design.

In addition to all of the Orgasmatron magic including free calling in the U.S. and Canada courtesy of Google Voice, you now get some terrific new features: randomly generated passwords for all of your extensions, free Skype support and a new backup module both of which we’ll introduce over the next few weeks. And CallerID Superfecta now is preconfigured to work out of the box with support from dozens of providers worldwide.

Among others, Ward Mundy (Nerd Vittles, our guest next week) and many of the VUC regulars join in this violent argument civil discussion about who is responsible for filtering, where it should take place and the how and why of their ideas on the subject.

If you’re into SIP technology, you’ll want to hear this discussion about who should protect people from SIP “CallerID stuffing” among Ward Mundy, Fred Posner (VoIP Tech Chat), Tim Panton, Karl Fife, Leif Madsen and the rest of the great gang of VoIP regulars. This is why you need to join us LIVE every Friday!

Programmers differ enormously over who should filter incoming data and where. There is no right answer, although the main point is to protect your users against whatever possible attacks might come through your system or pbx.

Martin Sunstrum, the founder of Aksys Networks joins us for a presentation of the KONNECT line of business VoIP phones.

He will give an overview of the KONNECT Business Phones and explore the tight feature integration with Asterisk-based PBX systems.

Lastly, he will present the intriguing KONNECT Office Phone System, a multi-site office phone system that requires no PBX or hosted service.

This weeks guest will be Michael Iedema, leader of the AskoziaPBX Project. AskoziaPBX is a lightweight embedded Linux + Asterisk distribution with a web GUI layer based upon the m0n0wall framework by Miguel Kasper.

It’s almost exactly a year since we last discussed AskoziaPBX. Earlier this week they celebrated the release of v2.0 after many months of work to migrate from FreeBSD to a Linux core. Michael will  be on-hand to discuss their recent efforts, the state of the software, growth of the developer team, project road map, etc.

We talk about several topics in part 2 including Skype, USB connections for VoIP, new devices like the Gigaset One, a cellphone gateway. Lots of talk about NAT with Asterisk and SIP. The usual chatter from friends around the table in our weekly “club”.

Among the participants Tim @steely_glint, Neil from IdeaSIP, Karl @karlfife, Michael @mjgraves and Graves on SoHo VoIP, David Frankel from ZipDX who furnish our great wideband conference facilities, Dave @DaveMichels from Pin Drop Soup, Pascal from Media5.

Welcome to a very old friend of mine, Bob from Future Video. The VUC encourages new people to be a part of our weekly talks. Please join us so you can chime in, ask questions and generally have a great time.

EC2 attacks continue with no help from them. Amazon continues its “head in the sand” approach to our community and this is unacceptable. Forgive the intrusion on this page, look below for the VUC sessions.

You can help push this to their attention:

Please make sure you keep this issue visible by voting it up on SlashDot. If you haven’t followed out discussions, see Fred’s story. Asterisk user mailing list has a lot of info on it as well. Post on Twitter, their robot stupidly repeats all comments that contain EC2 so don’t forget to use that mention in anything you post. Post on your blogs and any forums you can.

I expected better from Amazon and I’ll withdraw my significant business from them if they don’t rise up to the challenge.

[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-317307.mp3|titles=Kamailio 3.0]

Guests: Daniel-Constantin Mierla, Andrei Pelinescu-Onciul and Alex Balashov (Evariste Systems)

Discussion of why you would need Kamailio, who uses it, what is does.

Kamailio 3.0.0, the first release that combines the features of Kamailio (OpenSER) and SIP Express Router (SER) in the same SIP server instance, and the vision of future development within SIP Router project.

Kamailio 3.0.0:

- new features

- how to use it to deal with latest challenges in communications: SIP instant messaging, presence, and social networking

SIP Router

- project achievements since its start in 2008

- future development

Links from the IRC  #vuc channel

[audio:http://recordings.talkshoe.com/TC-22622/TS-333211.mp3|titles=pbx Choices]

A fight to the finish with VUC’s resident VoIP skeptic Dave Michels and Mike Oeth, CEO of  Junction Networks (OnSIP.com) on choosing between hosted and on-premises pbx.

What are the arguments for and against a hosted pbx for your business. Are they different for small and larger business pbx choices?

Dave and Mike fight it out with the help of referee Michael Graves and the roar of the crowd.

We will be welcoming Bria Product Director Todd Carothers to the VUC in a second segment on Jan 29th at 1PM EST.

From this press release:

CounterPath and Digium(R) Launch Asterisk-Based Multimedia Softphone

Based on open standards, Bria for Asterisk combines Digium’s Asterisk, AsteriskNOW and Asterisk Business Edition telephony platforms with Bria Professional, CounterPath’s enterprise-class, multimedia softphone. Bria for Asterisk gives users advanced telephony features, yet can be quickly set up and configured with any Asterisk deployment.

The VUC is officially testing Bria on Windows and OS X and we hope provide valuable feedback to Counterpath during this launch.

[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-315634.mp3|titles=Bria and Counterpath with Todd Carothers]

Yet another group of geeks discuss the iPad
[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-318318.mp3|titles=Yet another group of geeks discuss the iPad]

Direct mp3 link (part 1) – Direct mp3 link (part 2)

Part 1 [audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-256280.mp3|titles=Part 1 Skype for Asterisk]
Part 2 [audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-264537.mp3|titles=Part 2 Email servers SPF spam sendmail postfix]

Lots of talk about Skype for Asterisk, with Tim Panton who has been on the SfA beta for over a year and who will be speaking at Astricon in October.

IRC Transcript 2009-09-04

S675IP and an A58H accessory handset

Thanks to Gigaset of North America, we are planning a new giveaway of a Gigaset wideband-capable SIP/DECT phone.

To win the Gigaset you will have to be in the IRC channel #voip-users-conference as well as on the conference.

The winner will need to identify a short piece of music and the album it comes from.

Be there!

Coincidentally with the session on mobile voip with Ruben Olsen, I finally pulled the trigger on an iPod Touch “second generation”. This is IMO the iPhone without phone, camera or GPS. What that leaves is a very good mobile computing platform with WiFi, an mp3 and video player and the worlds hippest most intelligent interface to date.

NOTE: I will add new clients I test to this thread as comments

NEW note: Ruben just published this article on the subject.

Cutting to the chase, I have tried several SIP clients in the last 48 hours, and here are a few of my experiences.

The best performer has been iPico, $9.99 but far ahead of the others in most things. It will do only a single SIP account though. Call quality was best and it dials SIP URI, something none of the others do AFAIK. This is important because to dial in to Talkshoe, ZipDX and many other things that interest me, I need SIP URI. Edit: iSip has fixed the audio issue I was having and demonstrated excellent support in the process.

Next comes iSip, $6.99, (built on pjsip). Multiple SIP accounts, decent interface but a problem with outgoing audio quality.

The I tried WeePhone, $4.99. It has the least well done interface, but looked as if it would work fine, but the outgoing audio was absolutely unuseable, horribly distorted.

Skype (not really a SIP client, but…) works well but can not make free calls to SIP services so I’m just adding that I did test it and it does work. And it’s free.

Looking at the reviews on the app store isn’t too enlightening because many reviewers don’t know much about the technology, so they are reacting to what the interface looks like or the fact that they don’t know what auth user name or proxy is, which would impede them in getting the client to work.

The main trouble I have had so far is the microphone that is a part of Apple’s $30 headset. If I can figure out how to keep that facing me, I think things will work better

Olle E. Johansson (oej) is not just another boring developer, he’s a refreshing change, a family guy, a very good presenter and he shares our enjoyment of good wine. Olle has a vision and speaks of presence and the future of VoIP.

This video was shot with a tool I just got so there are mistakes in the switching. I’d rather leave it rough, including the audio dropping out, because the most interesting stuff is there. I promise to do better next time.

[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-206837.mp3]

IRC Transcript 2009-04-03

I’ve been testing all the ways I can find to connect between phones and Skype. I happened to be testing OpenSky when I saw some posts from Michael Robertson on the asterisk-users mailing list. As I exchanged a few emails with Michael, I realized he had been a founder of Mp3.com and well as Gizmo5.

He was generous enough with his time to spend a full hour (on a cellphone!) answering questions and sharing thoughts about the various SIP to Skype solutions, including Digium’s Skype for Asterisk, Gizmo’s OpenSky and Skype’s SIP for Skype which Michael calls vaporware.

Post conference chat, aka “Cognac”

The post conference has discussions about the expectation that all services should be free with David Frankel of ZipDX.com Randal Schwartz from FLOSS Weekly and the regular gang. Due to some problems with Talkshoe, various audio sources are used so you can hear what different phones sound like.

[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-210205.mp3]

[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-198844.mp3]

IRC Transcript 2009-03-27

SIP hacking has escalated and there was a wave of it in recent times. John Todd published a simple common sense set of rules that are worth reading. One of the main risk areas is using user names and secrets that are too easy to guess (such as extension 200, username=2000,password=wakeme).

More on SIP for Skype etc. That looks like it will be an ongiung discussion. See the session with Michael Robertson above.

We briefly reviewed a new book called Asterisk Gateway Interface 1.4 and 1.6 Programming. John Todd and /me have both been looking at it and we both think it looks good. Since the discussion, I’ve read more of it and I will talk more about it in a future session.

Amazon link (beer money!)
Asterisk Gateway Interface 1.4 and 1.6 Programming

We also mentioned AMOOCON where we can meet up for those of you who will be in Europe or are already there. I will be presenting two papers, one on VoicePHP and one on the VoIP community and various social networking tools.

[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-190012.mp3]

IRC Transcript 2009-02-20

Jason Fischl from Counterpath joins us to talk about their SIP clients and more.

[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-143738.mp3]

It’s down to the wire for Astricon, a few days. We talked to a few people who have experienced Nokia phones SIP stack. Other more general stuff, and the dCap certification process.

[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-137957.mp3]

Recent Voip News & Commentary, Digium partners with FreePBX, SIP/DECT vs ATA+DECT for cordless phones, Karle Fife on the new Rhino failover BRI/PRI card, John Todd on “The Proxifier”, Darrick Hartman announced Astlinux 0.6 expected next week, includes Asterisk GUI 2.0 Digital Opsys draw for Astricon 08 exhibitors pass Many thanks to Michael Graves for stepping in!

[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-130044.mp3]

SIP, “Today’s Hottest Protocol comes of age” says Ubiquity white paper. True? What more is there to say about SIP? We found plenty!

[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-91220.mp3]

Well known for high-quality SIP phones this presentation will be an overview of their SIP phones and explanation of their new SIP-DECT enterprise scale cordless technology.

[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-81881.mp3]

Junction Networks CEO Mick Oeth will present OnSIP, a “completely customizable sip network for hosted pbx, virtual pbx or any business communication need. OnSIP takes SIP to the next level, with free domain hosting, support for any SIP device, no user fees or seat licenses”.