News from Industry

Forecasts are overrated.

I’ve been asked time and again things related to the market sizing of WebRTC, and I’ve tried to shy away from it all the time. The Dilbert strip above explains why…

This whole notion of estimating the size of a market that is, to be frank, hard to define, without solid numbers, too new – all lead to the question: why bother?

What are you looking for? The market size of WebRTC contact centers? Is that only for the WebRTC piece of it? Greenfield ones? With or without call widgets in tiny WordPress sites? How do you place the monetary value on it? Is it the WebRTC part or the whole contact center you’re interested in? Do you want the number to amount to a billion $ and go backwards from there so it fits your desired strategy?

All useless.

2 billion users. X% CAGR. 15% YoY growth.

Sure.

Any day.

With something like WebRTC, such things are close to impossible as far as I can say, and probably not really worth it. Need to throw a number in the air? Generate it randomly.  It’s good enough for the TSA then why not for you?

Which leads me to something you won’t find in my WebRTC PaaS report – the one dealing with the WebRTC API market and assists developers in understanding if they should use a vendor and help picking up the right vendor (if that’s the course selected). These estimates don’t help in such a case. They are worse than useless.

Need estimates? Find some other report online. They will happily share their guesstimates in press releases out there (seen a few lately) so you can decide if it is worth paying for to get that “validation” you need for your management.

Need to make real decisions on how and what to implement? That probably won’t be in these reports.

The post You Won’t Find Guesstimates in My WebRTC PaaS Report appeared first on BlogGeek.me.

Losing customers because of issues with your network service is a bad thing. Sure you can gather data and try to prevent, but isn’t it better to prevent issues in the first place? What are the most common pitfalls to look out for? What’s a good benchmark? What WebRTC-specific user experience elements should you spend […]

The post The Big Churn – learning from real usage stats (Lasse Lumiaho and Varun Singh) appeared first on webrtcHacks.

There’s progress, but the real action will be in 2017.

There has been a lot of chatter lately around Apple’s snail-like progress in supporting WebRTC and Microsoft’s announcements at their BUILD conference. I am still left under-impressed but positive and confident. Here’s why.

Apple and WebRTC

Let’s start with Apple. The only official statement we will get from Apple will be “we have WebRTC”. Question is when, for what and if at all.

We have indications of progress in Apple, and Alex is keeping us updated on the goings with Apple and WebRTC.

I think Itay Rosenfeld  is making a good case why Apple needs WebRTC more than WebRTC needs Apple.

So we know WebRTC is of interest to Apple and we know it is being added to Safari.

We know one more thing. Apple is actually trying to refresh and update its Safari browser. Dare I say “modernize” it. They even recently started a Technology Preview for Safari, joining the rest of the gang of browser vendors to showcase their upcoming plans and intentions. That doesn’t include WebRTC, but WebKit indicates WebRTC as “in development” – and WebKit is the rendering engine used by Safari.

Will Safari include WebRTC? Yes.

When? My guess is end of 2016.

What will it include? WebRTC. H.264. No VPx “nonsense”.

Where? On Mac OS X, but not on iOS. That one will come in 2017.

Microsoft’s Romance with xRTC

Microsoft added ORTC to Edge. I shared my view about Edge already. To sum it up – great browser. No adoption.

To date, there has been little adoption of Edge/ORTC by vendors. If my memory serves me right, adopters include Twilio, &yet and Frozen Mountain. That’s less them impressive. And Microsoft knows that.

The problem here isn’t ORTC. It is Edge. And Microsoft seems to miss that minor detail.

At the recent Microsoft BUILD conference, a few announcement were made (thanks @hcornflower for the tip):

"We now have more than 150 million monthly active devices using Microsoft Edge" — @morris_charles #EdgeWebSummit pic.twitter.com/2q4ZVFeF9k

— Kenneth Auchenberg (@auchenberg) April 4, 2016

So. “150 million” monthly active devices. But no monthly minutes as in their last disclosure. I wonder what monthly active means and how many of them open it up just to get to IE when Chrome doesn’t work. I know that’s how I use it to get to a Silverlight site that my kid wants to use.

I guess this number was high and positive that the managers at Microsoft decided to focus on it instead of the more important number of average use time per user. This led them to this decision:

MS announces new WebRTC goodies coming to Edge:
H.264/AVC, VP8, MediaRecorder, DTLS 1.2, ECDSA certs pic.twitter.com/jDwRug2F13

— Justin Uberti (@juberti) April 4, 2016

1. Adding WebRTC and not ignoring it with an implementation of only ORTC. At long last, they got sense and decided to make it easier for developers to support Edge instead of having developers look at the abysmal market share of Edge. The challenge they face is that people who switch to Windows 10 opt for Chrome over Edge more often than not, and in the enterprise die hards stick with IE. Until this trend changes, there’s a real issue here
2. Supporting H.264, and not only their interanl H.264UC proprietary codec. This makes sense, as Chrome is adding H.264 and Firefox is already supporting it
3. VP8 is “under consideration”, so we will have it after H.264. Probably somewhere into 2017. Too late
4. No VP9. With current development speeds, I can see VP9 getting adopted en mass in many use cases and Microsoft Edge staying behind, with no Edge

It would have been better to just add WebRTC to IE11 in parallel than to entice users to switch altogether to Chrome.

When will vendors need to revisit Edge when it comes to WebRTC? Not before Q4 2016.

Microsoft Skype

Skype is interesting. Late to the market. 300 million active users. A lot, but unimpressive if you compare to the leading consumer communication services that are out there.

Skype for Web is what Google Hangouts did the first two or three years of WebRTC’s existence – took components of the WebRTC implementations, modified it to fit their needs and made a plugin for Hangouts out of it. Until they just made it “native” to the browser when they could.

As written in a recent comment I’ve read – they should have done this 5 years ago, but better late than never.

The more interesting part here is the newly minted Skype SDK. I think this is Skype’s third attempt at an SDK – there may have been more. Previous ones were failures. Not because of lack of adoption, but rather because the way developers were treated. This doesn’t bode well for this round. Especially not if you couple it with the current numbers and the size of Skype.

That said, I can easily see Lync/Skype for Business enterprises adopting the SDK to deal with customer support related requirements, taking a bit of the market from WebRTC PaaS vendors. To go beyond this use case, it will take more effort from Microsoft.

The Microsoft Skype for Web and SDK initiatives need to be viewed in the light of other players as well.

Cisco Spark

Cisco Spark (along with their Telepresence and UC offering) goes head to head against Lync/Skype for Business.

Cisco made several interesting moves lately:

• Acquired Tropo, to beef its Communication APIs and integrate them with Spark
• Acquired Acano, which fits nicely into high-end paying customers who use WebRTC
• Acquired Synata, to offer better search capabilities to Spark, to better compete with Slack
• Created the Cisco Spark Innovation Fund, and placed $150 million to developers building use cases on top of Spark

That’s a lot of milage to go against Skype for Web and the Skype SDK.

You can easily say that when it comes to publicizing and marketing their investment in communication services and enables, Cisco is ahead of Microsoft.

Google Hangouts

Google Hangouts is a shadow of what it can be when it comes to usage.

As a platform, it has it all. Everything you need to communicate, at a fraction of the cost of other solutions or for free. We use it daily at testRTC – both internally and to host meetings with customers and potential customers. We have no incentive to switch to anything else.

Hangouts adopted WebRTC from the beginning. First by embedding the WebRTC stack into the Hangouts plugin, using the components of WebRTC that it could, until it was able to just use WebRTC natively in Chrome. It still runs as a plugin on other browsers, but I assume that will change when WebRTC will be supported with all of the nuances of Hangouts.

What Hangouts is lacking is the traffic and the APIs to go along with its service. I am assuming Google are aware of it.

Apple FaceTime

Apple has FaceTime. Its proprietary service that should have been standardized at some point.

I’ll be surprised if Apple did anything interesting or serious when it comes to connecting FaceTime to WebRTC or adding an SDK to it. Or god forbid, let the poor people of the world who use Android – or a 5 year old Windows PC, connect to FaceTime.

Slack

Slack just added voice support with WebRTC and intends to add video. I’ve written about Slack a few times before, and how WebRTC is a logical investment for them. If they add integration points in their API that can access their real time communication capabilities it might become a very interesting player in the SDK/API space.

The real question in this case: Will a vendor using Slack continue using Skype in the long run?

Facebook and WhatsApp

Facebook Messenger uses WebRTC. WhatsApp somewhat uses it.

Skype has 300 million monthly active users. That’s way smaller than WhatsApp’s billion and Messenger’s 800 million. I am assuming there’s more voice and video calling happening on Skype on average per user than on either Messenger or WhatsApp, but the trend is probably towards Facebook and not Microsoft here.

The reason Facebook is so strong here is their new initiatives towards enabling businesses connect with their user base – the Facebook user base directly, which is the largest social network at the moment. If they want, they can throw in voice or video interactions with an SDK on top of it.

WeChat, LINE, ooVoo and Viber

All have integration points. All heading in multiple directions for monetization. Be it businesses connecting to their user base, market places, digital currency or bots.

Leveraging Skype as an SDK means you want their reach and users base. But all of these messaging plaforms have their user bases in the hundreds of millions of active users as well. They essentially compete over similar mind share and budgets of enterprises.

What’s in store for us in 2016?

More chatter and talks about Apple and Microsoft, but little in the way of progress by developers making use of Edge or Safari WebRTC capabilities. That will wait for 2017.

For Skype, there’s a challenge here, but also an opportunity. They can leverage WebRTC, focus on developers and come with use cases and success stories that will be hard to compete against. Microsoft is doing a lot already in this space, but there’s a lot more they need to be doing when you look at the competition they have.

 

Want to make the best decision on the right WebRTC platform for your company? Now you can! Check out my WebRTC PaaS report, written specifically to assist you with this task.

Get your Choosing a WebRTC Platform report at a $700 discount. Valid until the beginning of May.

The post Microsoft, Apple and WebRTC in 2016 appeared first on BlogGeek.me.

Messaging is used too much to stay only in the browser.

There seems to be a few conflicting trends going on at the moment:

• Most people on mobile consume their services through apps they install on the device
• People don’t install apps on their desktops and laptops anymore. They expect everything (most things?) to be available in the browser
• A new approach of Progressive Web Applications is on the rise, while at the same time, frameworks like React are becoming trendy
• Chrome dropping support for their app launcher due to limited use by users – at least from their own experience – people prefer launching their services from inside the browser than let it live on their desktop
• Mobile (and to some extent web) based messaging is leaving the browsers in favor of apps on desktops and laptops

This last trend is what I want to focus here. When all of the apps we use are now browser web apps on the PC, there are generally two types of apps I still install on my laptop:

1. Microsoft Office
   • I use Google apps whenever I can, but official documents to most of my customers still necessitate a Word file
   • Oh, and Powerpoint is a lot faster for me to create presentations with than the Google alternative
2. Developer tools
   • There’s something about development that still doesn’t fit in the browser
   • Oftentimes, I just need things locally or more responsiveness. Can’t explain why. Maybe I am just old fashioned

When it comes to communications, though, I prefer pinning tabs to the browser for the most common tasks I have – or just leave it to my phone. WhatsApp, Slack, Gmail – all get a pinned tab on Chrome for me. Whenever I need to use messaging in other domains (Facebook, LinkedIn, Meetup, Upwork, etc) – I just open a new tab in Chrome “on demand” and then close it once done.

I assume others install apps locally on Windows for things they want to use frequently. Which brings me to two interesting developments from the last year or so:

• Slack introducing a Windows app a year ago, essentially, it looks and acts like a glorified browser window dedicated to Slack
• Facebook introducing a Messenger app for Windows 10, which is just like their Messenger web app with the addition support for Windows 10 tiles

Great.

So we are now taking HTML5 web apps, wrapping them as Windows apps and install them locally.

It probably makes sense for a lot of the enterprise messaging apps – instead of just living inside the browser, be part of the installed set of apps on the desktop. Purists of WebRTC will complain that this is not how its done. Detractors of WebRTC will say it isn’t WebRTC at all. I’ll say it is just another way of using the technology.

If you want to take your own communication web apps and make a desktop application out of them, then the most popular approach these days that I know of is CEF – Chromium Embedded Framework. It takes your web app, and packages it with Chromium so that they both get downloaded and installed together.

I assume that this is what Slack used. I am not sure about the Facebook Messenger one though – the addition of Windows tiles is a complication, but probably solvable.

In a way, web and HTML5 have already took over our desktop. Even in apps what you get is HTML5 these days.

I wonder if and when will this trend hit mobile, and if so, will it be achieved via the new Progressive Web Apps approach.

The post Messaging is Migrating from the Browser to the Desktop appeared first on BlogGeek.me.

This week we had a number of awesome new features, but the most work went into adding translations to the verto communicator application! And we definitely couldn’t have done it without help from our wonderful community members! So far, we have translations for English, Spanish, Portuguese, Italian, French, Danish, German, Polish, Russian, Swedish, Indonesian, and Chinese. If you speak another language and would like to see that language available please submit a pull request with a translation!

Join us Wednesdays at 12:00 CT for some more FreeSWITCH fun! This week we have Chad Phillips talking about his FreeSWITCH Kickstart project! And, head over to freeswitch.com to learn more about FreeSWITCH support.

New features that were added:

• FS-8983 [mod_avmd] Enable on outbound channel to make debugging easier
• FS-8875 [mod_avmd] Enable faster beep detection
• FS-8688 [mod_vpx] Implement vp9 processing to avoid chrome hang
• FS-8972 [verto_communicator] Add i18n using angular-translate and static file loader
• FS-7125 [mod_sofia] Added an event “wrong_calls_state”. This is for fail2ban logging.
• FS-8990 [mod_verto] Adding verto_login header to verto::client_disconnect event
• FS-8991 [verto_communicator] Adding translations for French. Thanks Tristan Mahé
• FS-8989 [verto_communicator] Adding Portuguese i18n translations
• FS-8991 [verto_communicator] Adding French Canadian
• FS-8998 [verto_communicator] Adding German, Spanish, Catalan, Chinese, Polish, Russian, Swedish and Indonesian translations.
• FS-8992 [core] Indicate end of candidates in SDP to aid in the resolution of an interop issue with Mozilla
• FS-9000 [mod_conference] Set conference flags or conference member flags with individual variables per flag

Improvements in build system, cross platform support, and packaging:

• FS-8988 [mod_avmd] Rename files to include avmd in their name.
• FS-8971 [mod_amqp] There are two different status variables with two different meanings. This splits them back apart.
• FS-8933 [scripts] WIP Fix some breakage on raspbian as we don’t want the FS repos there yet because we don’t have armhf packages at this time

The following bugs were squashed:

• FS-8982 [core] Fixed an issue with play_fsv and play_yuv writing blank_img in parallel
• FS-8918 [core] Fixed an issue with a 10 Second timeout after Notify during Proxy refer
• FS-9002 [core] Fixed an issue with rtp timeout code parsing on video but its designed for audio
• FS-8749 [mod_conference] Fixed an issue when loading a video (mp4) for a conference using the “conference play” command “conference pause_play”
• FS-8993 [mod_av][mod_conference] Fixed a sync issue on conference playback for a video that is faster frame rate than the conference
• FS-8995 [verto_communicator] Added missing toastr in settings controller
• FS-8990 [verto communicator] Added verto_client_address to verto and presence events
• FS-8996 [verto_communicator] Fixed a typo in CAMERA_SETTINGS id and added some Italian translation
• FS-8997 [verto_communicator] Fixed fallbackLanguage
• FS-9012 [verto_communicator] Fixing sidebar in narrow resolutions clipping the video
• FS-8999 [mod_erlang_event] Fixed broken outbound connection
• FS-8913 [mod_sofia] Fixed a transfer issue when using bypass_media + SRTP + Inbound late negotiation
• FS-9004 [mod_http_cache] Set http get timeout on thread that is actively downloading with the value from the download-timeout configuration and added download-timeout param to prevent http_get from waiting unbound time for downloading to finish. Prevented prefetch threads from blocking if another thread is already downloading the same URL.

It’s been a while since last session about Kamailio during the weekly VoIP Users Conference moderated by Randy Resnick. Next Friday, on April 8, 2016, join the VUC session starting at 12:00 EDT (17:00 in London, 18:00 in Berlin, Madrid Paris, Rome) to get updates about the new features introduced by the latest major releases of Kamailio, what are the plans for future development and get a preview of what’s going to happen at Kamailio World Conference (May 18-20, 2016, in Berlin).Daniel-Constantin Mierla, co-founder of Kamailio project, will be there to give the updates and answer the questions from the audience.You can join the conference remotely via SIP or dialling one the many local PSTN numbers as well as listening via web or watching the video live streaming. Questions can be posted also via IRC channel #vuc on freenode.net. You can see more details about the ways to connect on the VUC website:

• http://vuc.me

Book the date in your calendar and join the session next Friday!

Not all of them.

Who is investing in its platform?

Twilio. Added a slew of services in 2015.

TokBox. Got a new Spotlight live broadcast service. But not only.

VoxImplant. Added HD to its audio conferencing.

The rest? Not really sure?

Most of the time, when people talk to me about their use case, and the need to pick a specific platform, it boils down to a shopping list of features. They want everything. Usually more than any single vendor can offer. When prodded further, they reduce the need to a small set of requirements. But then again, they do see in their future these added set of features.

In many cases, selecting a vendor means understanding which of them might have what you need in the future down the road in their roadmap – not necessarily in their service today, but they will get there by the time you will.

Guess what – this is another factor that needs to included to the list of requirements you need to look at when selecting a vendor to work with.

This is why in the latest release of my “WebRTC PaaS report”, I am adding a new section, which will give a quick indication to which vendors made changes to their platform (and if these changes were serious or not). The information there will date back two years, giving some perspective.

If you are thinking of stating to use one of the WebRTC API platforms out there and not sure which one to use, then this report may come in handy. Until this next updated release, I’ve taken the price down considerably – if you purchase now you pay $1250 instead of $1950 and you get a year of updates (so that the updated version will be yours next month the moment it gets published).

Check the WebRTC PaaS report page to decide if you need.

The post Which WebRTC PaaS Vendor is Investing in His Platform? appeared first on BlogGeek.me.

Novatel E371 (also known as Dell DW5804) is sold for less than $30 at Aliexpress, and it’s so far the cheapest 4G/LTE WWAN card suitable for PC-Engines APU.

The initialization is fairly simple, although it was tricky to find the right command (AT$NWQMICONNECT=,,).

cat >/etc/chatscripts/lte_on.E371 <<'EOT' ABORT BUSY ABORT 'NO CARRIER' ABORT ERROR TIMEOUT 10 '' ATZ OK 'AT+CFUN=1' OK 'AT+CMEE=1' OK 'AT\$NWQMICONNECT=,,' OK EOT cat >/etc/chatscripts/lte_off.E371 <<'EOT' ABORT ERROR TIMEOUT 5 '' AT\$NWQMIDISCONNECT OK AT+CFUN=0 OK EOT cat >/etc/network/interfaces.d/wwan0 <<'EOT' allow-hotplug wwan0 iface wwan0 inet dhcp     pre-up /usr/sbin/chat -v -f /etc/chatscripts/lte_on.E371 >/dev/ttyUSB0 </dev/ttyUSB0     post-down /usr/sbin/chat -v -f /etc/chatscripts/lte_off.E371 >/dev/ttyUSB0 </dev/ttyUSB0 EOT
Filed under: Networking Tagged: 3G, lte, pcengines, UMTS

Qualcomm Gobi 2000 is quite old (released 2009), but decent 3G modem, able to deliver up to 7Mbps in downstream in PPP mode. These modems in mini-pcie packaging are available at Aliexpress for less than $10, and make up a great option for 3G connectivity for PC Engines APU boards.

The modem needs a binary firmware to be loaded at the start. Numerous sources in Internet describe the ways to retrieve these files. The kernel driver in Debian 8 recognizes the modem as generic Qualcomm one, and sets up a QMI device (wwan0). But this model does not support packet mode, and you need to run PPP over ttyUSB1 device.

apt-get install -y gobi-loader wvdial mkdir /lib/firmware/gobi cd /lib/firmware/gobi wget --no-check-certificate -nd -nc https://www.nerdstube.de/lenovo/treiber/gobi/{amss.mbn,apps.mbn,UQCN.mbn} cat >/etc/wvdial.conf <<'EOT' [Dialer Defaults] Init1 = ATZ Init2 = AT+CGDCONT=1,"IP","internet" Phone = *99# New PPPD = yes Modem = /dev/ttyUSB1 Dial Command = ATDT Baud = 9600 Username = '' Password = '' Ask Password = 0 Stupid Mode = 1 Compuserve = 0 Idle Seconds = 0 ISDN = 0 Auto DNS = 1  EOT cat >/etc/network/interfaces.d/ppp0 <<'EOT' auto ppp0 iface ppp0 inet wvdial EOT

Also this script is useful for 3G connections, because with some providers, the Internet connection gets stalled every few days and needs to be re-connected.

Filed under: Networking Tagged: 3G, GSM, linux, pcengines, UMTS

San Francisco, CA - April 1st, 2016 - With skyrocketing demand for integrations between different services, 2600Hz is staying ahead of the competition with a unique integration named “Carrier Pigeons”.  This service allows customers to place calls to a specific number/extension where you then leave a voicemail.  2600Hz seamlessly transfers the voicemail to a thumb drive. Via automated, scalable, elastic engineering robots, the thumb drive is attached to a pigeon who is dispatched automatically to the specific location determined by the extension.  Using 2600Hz Mobile services, Carrier Pigeons are tracked via GPS until they reach their destination. Their progress is shown in an amazing user interface, the Monster Pigeon app.

2600Hz’s Co-Founder Darren Schreiber explains “We’ve seen what employing contract workers has done for the transportation industry, with services like Uber and Lyft. We see no reason why this can’t be adopted in the communications industry. The challenge in our industry is that people want their voicemail messages quickly - faster than a person or a car can deliver the message. We realized that Carrier Pigeons were a natural choice.“

Co-Founder Patrick Sullivan expanded on the project further: "We are always on the bleeding edge.  Now that we have integration with Voice, Video and SMS/MMS we had to think outside the box.  When sitting in our conference room talking about the future of business communication, we noticed a vast amount of resources sitting outside our window – Pigeons.  Most see dirty animals with wings. We see a way to deliver secure messages to places that might not have fiber or any other transit to get information to.  We also believe everyone has a crazy uncle or aunt ‘living off the grid’ and you might want to communicate with them.  We now have a way to do just that. This gives us a huge advantage over the typical CLEC/ILEC infrastructure.  The opportunities are endless.”

The program is in Beta and is invite only. Project “Carrier Pigeon” will launch publicly in the next couple months.  2600hz has already received multiple Silicon Valley investment offers and hopes to display the technology live, on stage, at TechCrunch, with other successful projects such as Pied Piper.

For questions and information requests please contact:

Captain Crunch
CaptainCrunch@2600Hz.com
140 Geary St.
San Francisco, CA
Ph: 415-886-7900

The FreeSWITCH 1.6.7 release is here! This is a routine maintenance release and the resources are located here:

Release files are located here:

• Tarball: http://files.freeswitch.org/releases/freeswitch/freeswitch-1.6.7.tar.gz
• Packaging: https://freeswitch.org/confluence/display/FREESWITCH/Installation (Please Note the updated location for 1.6 packages)

New features that were added:

• FS-8908 [verto communicator] Link to preview camera and microphone under settings
• FS-7800 [verto communicator] Added support for calling extra screens with same extension as the original and place the parameter conferenceCanvasID with the desired canvas id into the call parameters in the same place bandwidth preferences are added
• FS-8972 [verto_communicator] Fixed mute video translation and added i18n using angular-translate and static file loader
• FS-8989 [verto_communicator] Adding Portuguese i18n translations
• FS-8991 [verto_communicator]Adding translations for French and added French Canadian. Thanks Tristan Mahé
• FS-8998 [verto_communicator] Adding German, Catalan, Polish, Russian, Swedish, Chinese, and Indonesian translations.
• FS-8990 [mod_verto] Adding verto_login header to verto::client_disconnect event
• FS-8735 [mod_sofia] Display update support for Panasonic devices
• FS-6833 [mod_sofia] Allow FreeSWITCH to initiate late offer calls
• FS-8879 [mod_sofia] Work on SIP UPDATE and attended transfer for IPv6
• FS-8964 [mod_sofia] Make it possible to disable picture_fast_update INFO requests to de-clutter trace and logs
• FS-7125 [mod_sofia] Added an event “wrong_calls_state”. This is for fail2ban logging
• FS-8864 [core][mod_av] Improve video file playback
• FS-8977 [mod_av][core] Add functionality to support the NVENC hardware encoded h264 in codec and files and  set default to enable hardware encoder on conference too
• FS-8862 [core] Auto adjust on passthru
• FS-8876 [core] Add CPU affinity to each video thread in a round robin fashion.
• FS-8909 [core] Add feature to play background video while recording inbound video. This feature is suitable to provide some kind of feedback like an animation of a glowing record light etc.
• FS-8870 [core] Add human-readable call quality statistics logs on call hangup
• FS-7776 [mod_kazoo] Integrate the module into build system
• FS-8737 [mod_kazoo] Add required variables to default filter
• FS-8595 [mod_conference] Improve auto bitrate in personal canvas mode and do not let auto bitrate exceed native picture size
• FS-8921 [mod_conference] Re-designed the banner code
• FS-8685 [mod_conference] Multiple member arguments for conference related API command. The new format is: ‘conference foo relate 1 2,3,4 nohear’ or ‘conference foo relate 1,2 3 nospeak
• FS-9000 [mod_conference] Set conference flags or conference member flags with individual variables per flag
• FS-8822 [mod_callcenter] Added a real-time counter for calls in a queue.
• FS-8903 [mod_local_stream] Add a logo image and some parameters to control its behavior
• FS-8932 [mod_managed] Add in process load in-process plugins
• FS-8688 [mod_vpx] Add more detailed logs and tweak some parameters for the coming libvpx 1.5.0 and implement vp9 processing to avoid chrome hang
• FS-8728 [mod_amqp] Adding logging profile and functionality and added default configurations
• FS-8971 [mod_amqp] There are two different status variables with two different meanings. This splits them back apart.
• FS-8983 [mod_avmd] Enable avmd on outbound channel to make debugging easier
• FS-8875 [mod_avmd] Enable faster beep detection

Improvements in build system, cross platform support, and packaging:

• FS-8111 [mod_sofia] ‘sofia’ API command auto-complete cleanup
• FS-8763 [mod_sofia] Changed to set is_auth only after the results for switch_ivr_set_user
• FS-8782 [mod_verto] Fixed build error for missing __bswap_64 on Solaris
• FS-8778 [mod_verto] Define __bswap_64 for FreeBSD as well
• FS-8195 [src] Allow Solaris privileges to work on both Solaris and derivatives
• FS-8776 [build] Support GNU make parallel builds
• FS-8942 [build] Pass compiler to libvpx configure to account for environments without gcc
• FS-8871 [configuration] Fixed encoding “&” and “<” symbols in vanilla configuration
• FS-8808 [core] Fixed ^D in fs_cli with editline to delete char under cursor, not just backspace
• FS-8818 [core] Refactor X-PRE include to not toss error if there are no files that match the include.
• FS-8992 [core] Indicate end of candidates in SDP to aid in the resolution of an interop issue with Mozilla
• FS-8788 [Debian] Enabling stacksize limits for Debian packaging
• FS-8933 [Debian] Basic FreeSWITCH from source installer that works on Raspian and Debian. Also installs VertoCommunicator and LetsEncrypt SSL Certs. LetsEncrypt requires the machine to have a public IP and DNS for the FDQN functioning properly in public DNS
• FS-8948 [Debian] Handle non-existent configuration in Debian postinst causing failure to copy vanilla configuration on installation
• FS-8841 [Debian] Prevent hanging by pushing events to another queue to be processed in a different sofia thread.
• FS-7389 [CentOS] Correct the location of the freeswitch users homedir in the specfile
• FS-8186 [Windows] Fixed build with –disable-libyuv to ensure projects don’t incorrectly depend on libapr directly
• FS-8952 [Windows] Fixed an issue with unreachable code preventing build and simplify conditions
• FS-8779 [mod_shout] Properly detect lame/lame.h for FreeBSD
• FS-8808 [mod_avmd] Code refactor to fix indentation and conditional loops
• FS-8961 [mod_avmd] Add optional requirement of consecutive streak of estimations in SMA buffer to avoid false positives, fix definitions, and add options to control debugging/printing
• FS-8988 [mod_avmd] Rename files to include avmd in their name.
• FS-8878 [mod_amr] Fixed compiling without the library installed
• FS-8756[mod_say_nl] Improve dutch localisation
• FS-8933 [scripts] WIP Fixed some breakage on Raspbian as we don’t want the FS repos there yet as we don’t have armhf packages at this time

The following bugs were squashed:

• FS-8719 [mod_conference] Fixed a segfault caused by building without video support, but specifying video_mute_png variable for a conference member
• FS-8752 [mod_conference] Fixed a pixelation issue in initial seconds of recording a conference
• FS-8354 [mod_conference] Fix for G722 audio issues
• FS-8938 [mod_conference] Clear res_id when setting the same res_id to another member
• FS-8951 [mod_conference] Video lockup in conference due to race condition
• FS-8957 [mod_conference] Fix for video image blipping on personal canvas mode when 1 participant is watching video on hold
• FS-8750 [mod_conference] Fixed a set but not used variable warning
• FS-8749 [mod_conference] Fixed an issue when loading a video (mp4) for a conference using the “conference play” command “conference pause_play”
• FS-7915 [mod_sofia] Parse and store multiple path fields
• FS-8663 [mod_sofia] Added saftey checks for ;fs_path= command
• FS-8759 [mod_sofia] Fixed a segfault caused by device or provider timing interactions
• FS-8879 [mod_sofia] Fixed SIP UPDATE and attended transfer for ipv6
• FS-8898 [mod_sofia] Log setVariable at debug to easier tell what variables are being set from scripts
• FS-8913 [mod_sofia] Fixed a transfer issue when using bypass_media + SRTP + Inbound late negotiation
• FS-8796 [mod_verto] Fixed and cleaned up the mcast code
• FS-7132 [mod_verto] Fixed an issue with multiple websockets overwriting the jsock hash entry for a given sessionid causing a disconnection with the verto javascript application.
• FS-8842 [verto] Fixed an issue where calls created using the originate command lose audio when left on hold for 45 seconds
• FS-8877 [verto] Fixed an issue caused by Chrome Canary removing some audio mandatory constraints that break Verto
• FS-8805 [verto] Added a sanity check on array
• FS-8945 [verto_communicator] Don’t show preview_settings window during a video conference call or you will lose conference video stream until you refresh the page
• FS-8842 [verto] Fixed an issue where calls created using the originate command lose audio when left on hold for 45 seconds
• FS-8877 [verto] Fixed an issue caused by Chrome Canary removing some audio mandatory constraints that break Verto
• FS-8839 [verto_communicator] Fixed screenshare not displaying in the proper conference when caller is transferred to another conference.
• FS-8963 [verto_communicator] Fixed the status call label
• FS-8966 [verto_communicator] Fixed the ability to cancel dialing while doing speed test
• FS-8970 [verto_communicator] Set useSpeak/useMic/useCamera to ‘none’ when calling a secondary canvas
• FS-8765 [verto_communicator] Make the preview button work properly
• FS-7800 [verto_communicator] Disable video floor changes on multi-canvas
• FS-8995 [verto_communicator] Added missing toastr in settings controller
• FS-8990 [verto communicator] Added verto_client_address to verto and presence events
• FS-8996 [verto_communicator] Fixed a typo in CAMERA_SETTINGS id and added some Italian translation
• FS-8997 [verto_communicator] Fixed fallbackLanguage
• FS-9012 [verto_communicator] Fixing sidebar in narrow resolutions clipping the video
• FS-8720 [core] Fixed a segmentation fault when switch_channel_str2cause is called
• FS-8960 [core] Reset the buffer position to zero to ensure correct calculation of average
• FS-8975 [core] Fixed the dtmf_type and sofia profile parameter dtmf-type variables
• FS-8731 [core] Fixed a crash when leg-b invite video in voice call
• FS-8734 [core] Cleaned up video jitter buffer by adding some formatting to the debugging logs so the text jumps around less and fixing sequence number rollover code to handle rollover better.
• FS-8713 [core] Fixed a crash on bad video rtp stream by pushing a patch to make the sizes match. This was the original intention since we want to preserve the packet as-is while in the jitter buffer
• FS-8721 [core] Fixed an eavesdrop memory leak caused by moving bug_remove_all after destroy where it’s more than safe to kill bugs indiscriminately
• FS-8673 [core] Fixed a core dump on playback after “Decode Codec is not initialized!” log message
• FS-8571 [core] Add missing ENABLE_SRTP ifdef to allow building without SRTP
• FS-8770 [core] Fixed media_bug_answer_req=true creating a file for unanswered calls
• FS-8786 [core] Fixed a fax hangup problem when testing long faxes over T.38 and rtp-timeout-sec is set to 300
• FS-8802 [core] Fixed an issue caused by the sent timestamp of RTP rolling over to 0 and hitting an error condition causing it to stop sending audio
• FS-8726 [core] Fixed a spurious case of a thread stuck and saturating CPU
• FS-8812 [core] Respect in_thread_only parameter in switch_channel_check_signal()
• FS-8809 [core] Fixed an undeclared MAP_POPULATE causing compliation failure
• FS-8816 [core] Fixed switch_core_hash_insert_destructor not checking switch_hashtable_insert_destructor() returns
• FS-8830 [core] Added SDP line separator fix for SDP generated by the core. Sofia when giving an SDP through SOA was parsing the SDP and rewriting it. For endpoints that use the core SDP as is will have more accurate SDP now.
• FS-8806 [core] Change group_confirm_cancel_timeout to apply only to the legs that answer the call to prevent the remaining legs ringing indefinitely if the answered leg hangs up while executing the group confirm script.
• FS-8821 [core] Fixed switch_core_hash_insert_locked and switch_core_hash_insert_wrlock to make sure they check the status of the operation before returning a switch_status.
• FS-8862 [core] Auto adjust on passthru
• FS-8811 [core] Fixed an intermittent crash
• FS-8851 [core] Fix for codec for recording negotiated before call answered
• FS-8904 [core] Fix memory leak in img_write_text
• FS-8905 [core] Fixed for a heap buffer overflow detected by ASAN in the img patch/img overlay and it may show up as a rare segfault when using video with banners
• FS-8864 [core] Set video ready on first push to avoid catch 22 on some video files and improve video file playback
• FS-8910 [core] Properly negotiate SDES when receiving an SDP with a=crypto:0 with the wanted crypto suite, we should maintain that crypto tag in the local SDP in order for SDES setup to succeed.
• FS-8914 [core] Fixed mp4 recording cutting off the end in some cases by adding code to mitigate the sync of the end of the file encoding to add some padding to the end
• FS-8761 [core] Fixed a memory leak
• FS-8855 [core] Fixed the calculation of variance of tone’s frequency estimator to prevent flawed audio
• FS-8928 [core] Flag a binding error when using EventConsumer::bind with invalid event name instead of blindly using custom
• FS-8168 [core] Use copy image functions from libyuv instead of our home rolled versions as the libyuv versions have optimizations
• FS-8982 [core] Fixed an issue with play_fsv and play_yuv writing blank_img in parallel
• FS-8918 [core] Fixed an issue with a 10 Second timeout after Notify during Proxy refer
• FS-9002 [core] Fixed an issue with rtp timeout code parsing on video but its designed for audio
• FS-8847 [core] When a call ends sometimes the RTP stack complains of a bad poll. This is normal caused by shutting down the rtp on hangup. Add some code to catch the condition that is ok and still print the error otherwise
• FS-8789 [mod_conference][mod_av] Fix warning thats printed when it shouldn’t be and remove ability to swap to personal canvas while recording and prevent recording while personal canvas is on.
• FS-8864 [mod_av] Fixed regression to recording to improve file playback
• FS-8868 [mod_av] Set recording app to respect bandwidth set in SDP
• FS-8916 [mod_av] Fixed an issue with newer x264 library returning Encoding Error -1 for newer x264 libraries
• FS-8836 [mod_av] Fixed a deprecated warning on newer ffmpeg to support ffmpeg 2.8 and 3.0 in addition to 2.6
• FS-8914 [mod_av] Adding code to mitigate the sync of the end of the file encoding to add some padding to the end
• FS-8973 [mod_av] Enable libavcodec experimental native implementation features (such as AAC) by default
• FS-8959 [mod_av] Refactor to improve memory processing and prevent leaks in encoding h264
• FS-8993 [mod_av][mod_conference] Fixed a sync issue on conference playback for a video that is faster frame rate than the conference
• FS-8656 [src] Fix switch_event_channel_broadcast identifier redeclared
• FS-8777 [mod_redis] Add the missing netinet/in.h include required to build mod_redis/credis.c for FreeBSD
• FS-8768 [mod_callcenter] Releasing database handle after reserving agent
• FS-8856 [mod_callcenter] Updating a member fails because the agent_dispatch_thread removed the member just before it tried to update it.
• FS-8838 [mod_rayo] Improve logging and error handling when executing API on output component. Attempt to make output component stop a little more robust and do a better job of detecting when output component completed because of hangup and preventing operations on output component when call has ended.
• FS-8811 [mod_local_stream] Fixed an intermittent segfault and a crash caused by dividing by 0
• FS-8866 [mod_erlang_event] Fixed memory leaks caused by not destroying session event_hash and events in queue
• FS-8999 [mod_erlang_event] Fixed broken outbound connection
• FS-8871 [configuration] Fixed encoding “&” and “<” symbols in vanilla configuration
• FS-8943 [configuration] Fixed misspellings in two comments
• FS-8884 [build] Add –disable-libyuv and –disable-libvpx configure arguments to disable building these libraries on FreeBSD and smartos
• FS-8915 [mod_smpp] Shortened event header name
• FS-8950 [mod_skinny] Fixed a few memory leaks
• FS-8946 [mod_xml_cdr] Fixed a segfault on call after loading with no config file or event bind failure causing the module load to fail
• FS-8937 [mod_easyroute] Handle a segfault when using bad customer query or on query error
• FS-8775 [mod_say_de] Add function SST_SHORT_DATE_TIME to mod_say_de.c and some language tweaks
• FS-8406 [mod_rtmp] Add options to scale down cavas size, fps, and bandwidth
• FS-8736 [spandsp] Fixed missing MEMMOVE macro in spandsp autoconf
• FS-8891 [RTP] Fix for T38 fax failing between 2 freeswitch boxes with high cpu usage
• FS-8312 [mod_sangoma_codec] Some maintenance clean-up and reset the whole trans-coding session memory on destroy to make debugging easier
• FS-8971 [mod_amqp] Resolve global structure handling issue causing a core dump. Thanks to Ben Hood for reporting the issue.

March 30, 2016: Kamailio v4.4.0 is out –  a new major release, bringing out new features and improvements added during about seven months of development and one and a half month of testing.In short, this major release brings 9 new modules and enhancements to more than 60 existing modules, plus components of the core and internal libraries. Detailed release notes are available at:

• https://www.kamailio.org/w/kamailio-v4-4-0-release-notes/

The release series 4.4.x is expected to be the last starting with number 4, next major release is planned to be 5.0.0, which should get Kamailio to a new era — there are already ongoing discussions on users mailing list about the changes to be done as well as a wiki page was created to collect the relevant ideas for 5.0.x. Join us to design and develop v5.0.0!Enjoy SIP routing in a secure, flexible and easier way with Kamailio v4.4.0!Thank you for flying Kamailio and looking forward to meeting you at Kamailio World Conference 2016!

This week we have added functionality to support the NVENC hardware encoded h264 codec, files, and default on conference.

Join us Wednesdays at 12:00 CT for some more FreeSWITCH fun! And, head over to freeswitch.com to learn more about FreeSWITCH support.

New features that were added:

• FS-8964 [mod_sofia] Make it possible to disable picture_fast_update INFO requests to de-clutter trace and logs
• FS-8977 [mod_av][core] Add functionality to support the NVENC hardware encoded h264 in codec and files and  set default to enable hardware encoder on conference too

Improvements in build system, cross platform support, and packaging:

• FS-8186 [Windows] Fixed build with –disable-libyuv to ensure projects don’t incorrectly depend on libapr directly
• FS-8961 [mod_avmd] Add optional requirement of consecutive streak of estimations in SMA buffer to avoid false positives, fix definitions, and add options to control debugging/printing

The following bugs were squashed:

• FS-8960 [core] Reset the buffer position to zero to ensure correct calculation of average
• FS-8975 [core] Fixed the dtmf_type and sofia profile parameter dtmf-type variables
• FS-8312 [mod_sangoma_codec] Some maintenance clean-up and reset the whole trans-coding session memory on destroy to make debugging easier
• FS-8963 [verto_communicator] Fixed the status call label
• FS-8966 [verto_communicator] Fixed the ability to cancel dialing while doing speed test
• FS-8970 [verto_communicator] Set useSpeak/useMic/useCamera to ‘none’ when calling a secondary canvas
• FS-8765 [verto_communicator] Make the preview button work properly
• FS-7800 [verto_communicator] Disable video floor changes on multi-canvas
• FS-8914 [mod_av] Adding code to mitigate the sync of the end of the file encoding to add some padding to the end
• FS-8973 [mod_av] Enable libavcodec experimental native implementation features (such as AAC) by default
• FS-8959 [mod_av] Refactor to improve memory processing and prevent leaks in encoding h264
• FS-8836 [mod_av] Fixed a deprecated warning on newer ffmpeg to support ffmpeg 2.8 and 3.0 in addition to 2.6
• FS-8750 [mod_conference] Fixed a set but not used variable warning
• FS-8971 [mod_amqp] Resolve global structure handling issue causing a core dump. Thanks to Ben Hood for reporting the issue.

No easy answer.

What route should your messaging implementation take?

If there’s something I like is to write code. I haven’t done so in years, but it still is my passion. A year or two ago, I’ve done a small coding project for something I needed. After a whole day of coding it dawned on me that I haven’t checked my email, social networks or notifications the whole time – and didn’t even miss it. The only thing these days that can focus me on a single task at a time is programming.

When I did develop, and manage developers, there was always that tension of NIH in the air – the Not Invented Here syndrome that we developers are so good at. We want to develop stuff on our own and not “outsource” it to others. Hell – if I wrote a piece of code a year ago it was crap the next year and had to be rewritten.

I had the chance to listen in to Apigee’s recent webcast on Build vs Buy API Management. See it here:

This webcast goes over a lot of reasoning I see going on in any development project when the decision needs to be building build and buy.

The funny thing is that I don’t hear this kind of a discussion enough when it comes to messaging. Somehow, people think it is trivial.

I took a few of the concepts in this webcast, and “translated” them into the realm of build vs but for messaging.

Limited view of the scope

When a project starts, it seems that adding messaging isn’t that hard. You have a bunch of people. Maybe some presence indication. Run around a few Websocket messages for the text involved in the conversation and you’re done.

But is it really true, or is there more to messaging? It is far from trivial. Even simple things like delivering messages while disconnected or handling push notifications are notoriously hard to get right – even for those who should be the experts in it.

When you define what it is you need to build for your messaging, most often than not, you’ll be doing it with the following “mistakes”:

• You will have a narrow scope of what is really needed
• You will focus on the functional part of messaging, but probably a lot less of the other requirements (such as a good backend to understand what your system is doing and how people end up using it)

With limited scope comes the challenge of not comparing the right things when deciding between build or buy.

RISK

Every development project is risky. Purchasing an off the shelf solution usually mitigates the risk by having it done by someone else where the payment and deliverables are known in advance.

Developers tend to ignore risk – especially if the project is interesting enough to build. And yes. A distributed, low latency, high efficiency, large scale messaging backend written in Lua or Go is highly interesting.

You are not WhatsApp. Or Netflix

Building your own messaging system is hard. It takes a lot of effort. WhatsApp seems so easy, but getting there is hard.

This shift towards in-app messaging that is occurring means that in most cases, messaging is becoming part of an IT project and not exactly an R&D project. As a company, this means the focus is elsewhere and that messaging is considered a commodity or a non-core technology.

In such cases, there is no real funding for ongoing development, support and maintenance of an in-house DIY messaging framework.

Can open source help?

Sure, but is it at the right level of maturity?

There are a few dozen open source messaging frameworks out there. They probably do the work, but barely.

And the main challenge is that messaging is rapidly changing, which means that whatever is out there today is probably somewhat obsoleted or out of sync with what you need anyway – and getting it to where you need it means more investment on your end. Probably.

To top it all, with most of these open source initiatives, what you’ll find out that they have one main contributor behind them. That contributor is most probably a vendor who is offering support and proprietary modules to take care of commercializing the open source offering. Things like reporting, scaling, maintenance, etc. – all these will fall in the domain of proprietary and payment.

So if the idea from the start was to use open source to refrain from having to negotiate and work with a vendor, where does that lead you down the road? Isn’t it better to acknowledge the fact from the onset and find a suitable solution out of a larger set of available vendors?

Time To Market

I know. I know.

If you write your own messaging system, it will take you the better part of a weekend. Adding a bit of code and stability around it clocks it at a month. Nothing can beat that.

But what is it you are comparing here? Are you concerned about your prototype implementation or is that like production grade we’re talking about?

Getting something to production requires a lot more time.

Why are you even going DIY?

Is it because it will be cheaper?

Because you’ll have more control over your future and destiny?

DIY is going to cost you in time and effort which you don’t necessarily have.

If and when this project of yours going to succeed, you’ll find out that with it more requirements and maintenance work is necessary. But what you’ll also find out is that the budget might not be there for you to handle that extra load in development. You promised the organization a working messaging system, and now that it is working – why are you asking for more funding exactly?

 

 

Easy? Hard? Core? Commodity?

I guess in most cases, deciding to develop your own messaging system requires a very good reason.

At testRTC we had that same need, though slightly different. We needed a way to communicate with the browser machines we’re running. It was all fine and well when the number of machines was rather small and their locations were simple. It became a real headache when we grew bigger and when customers started connecting machines in locations with flaky internet connections. We ended up using integrating one of the realtime messaging players for that purpose – and haven’t looked back at it since.

Messaging might seem easy, but it is pretty hard once you get to the details.

So why not outsource it and be done with it?

The post DIY or SaaS for Your In-App Messaging? appeared first on BlogGeek.me.

Here we are, the 9th edition of Kamailio Awards granted for the activity related to Kamailio and Real Time Communications during the previous year, respectively 2015. Continuing the tradition, there are two winners for each category.

During 2015, Kamailio v4.3 was released and most of the development for upcoming v4.4 was done  (to be released on March 30, 2016). The 3rd edition of Kamailio World Conference was organized, gathering the community of developers and enterprises relying on Kamailio and its ecosystem.

The 2016 is going to be a reference year in the evolution of the project. It marks 15 years of the development for Kamailio project, maybe not a smooth path always, but with amazing results after all these years. We will celebrate it at the 4th Kamailio World Conference, during May 18-20, 2016, in Berlin, Germany.

More of 2016 - after the release of v4.4, the resources will be allocated to design and bring Kamailio to the next level of flexibility and scalability, a project already code-named Kamailio 5.0.

To cut it short here, expect an amazing evolution of Kamailio in the near future. Now back to the awards.

Next are the categories and the winners!

New Contributions

• http client modules - http_client and http_async_client - split development between Olle E. Johansson, Hugh Waite, Federico Cabiddu and Camille Oudout - the two modules were developed separately and by that it shows that interactions with external applications via HTTP API are very important for building modern telephony systems.
• tlsf (two-level segregate fit) memory manager - a core component developed by Camille Oudout (from Libon, Orange, France), bringing in an alternative to the existing memory managers, which relies on a modern management algorithm for memory operations (alloc, free and join) that is more suitable for handling various special cases of SIP server deployments with large number of TLS connections. This contribution "forced" another important enhancement: the addition of the command line option to select the desired memory manager at Kamailio startup.

Developer Remarks

• Hugh Waite (Xura, UK) - a long term developer of Kamailio, besides the contributions to HTTP client implementation already mentioned above, he was very active on adding new features or tracking issues to many other modules, such as websocket (adding support for SIP fragmentation), pv, tm, utils, app_lua, json, sdpops, a.s.o.
• Stefan Mititelu (1&1 Germany) - he had a consistent amount of commits to various components, including support for database management of a RTPEngine farm, enhancements to debugger module to print the new SIP message after the config changes, refactoring of tm statistics, per module memory usage summary, a.s.o.

Advocating

• Dragos Vingarzan - the initial author of IMS module in Kamailio (developed via OpenIMSCore), Dragos helped the project to organize many developer and community meetings at FhG Fokus and continue to promote the project via Core Network Dynamics company.
• Simon Woodhead - a long time player in the VoIP and SMS businesses with Simwood eSMS Ltd, Simon has been a promoter of Kamailio, presenting at various events and publishing articles about interesting topics such as VoIP fraud or building mobile services.

Technical Support

• Emmanuel Schmidbauer - besides several patches and activity on the mailing lists, Emmanuel was very active on #Kamailio's IRC channel, answering to and guiding the participats to solve their issues in real time.
• Mikko Lehto - active in Kamailio community for several years, answering on mailing lists, Mikko had also a consistent contribution by compiling Kamailio on different operating systems, pushing patches that cleaned up a lot of specific warnings.

Blogging

• Gholamreza Sabery - for publishing the Ansible playbooks and related files for an Active-Passive Kamailio auto-deployment using Pacemaker and Corosync.
• blog.irontec.com - for publishing an ample tutorial about horizontal scaling of VoIP platforms using Kamailio and Asterisk with Docker (in Spanish, but configs and commands can be taken directly, text can be easily translated with web tools).

Related Projects

• matrix.org - an admirable effort led by Matthew Hodgson to specify a protocol and build an open source platform to connect heterogeneous RTC systems, with a connector for SIP and Kamailio. Given the trend of big companies to build walled garden RTC services, Matrix is aiming to provide the framework that will make it easier to interconnect, relying on extensible technologies such as WebRTC and HTTP/JSON APIs.
• sngrep - the swiss army knife of sniffing and analyzing the SIP traffic on a terminal (e.g., when connected via ssh). It can draw diagrams of SIP dialogs (VoIP calls) with updates in real time, it also provides advanced match, search and sort criteria. If you haven't used so far, think of it like a wireshark for SIP traffic in the terminal.

Business Initiatives

• Pascom, Germany - the vendor of Mobydick - an Asterisk-based PBX system - with a cloud service leveraging Kamailio for security and scalability
• VoiceTel, USA - an IP telephony operator in North America, using Kamailio to take care of filtering bad and good traffic and load balance across media servers farms

Events

• IIT RTC Conference - one of the few events out there that still tries to cover a broad range of RTC topics, with a tight relation to research and academic environments, and a particular focus on future needs of communications.
• TAD Hack - Telecom Application Developer Hackaton - started and mainly organized by Alan Quayle, it is a series of events along year long taking place in different locations across the world, even allowing remote participants. The events try to promote innovation in telecom space by getting together developers of different applications, platforms and services. Developers and friends of Kamailio are often participating, among them Carsten Bock, Federico Cabiddu, Giacomo Vacca, James Body, Randy Resnick.

Friends of Kamailio

• Carol Davids - Professor at Illinois Institute of Technology, Chicago, USA - under her supervision, many students of the Master program at IIT are designing and developing new concepts for RTC using open source applications, involving Kamailio in many cases. Actually IIT has been awarded in the past as an academic entity, due to a number of research papers that were published by their members, where Kamailio has been used during the proceedings.
• Markus Monka - Head of Infrastructure IT/Telco at sipgate, Germany - an early adopter of SER, continuing with OpenSER and Kamailio, sipgate is known as one of the companies pioneering the VoIP services for residential and business customers, serving several hundred thousands of active connections. Markus has been with the company for more than 10 years, supporting and assisting Kamailio to organize many of its events during the past years. He coordinates the team that contributed many patches and documentations back to the projects.

This is it for 2014. If you want to check the previous turn of awards, visit:

• http://blog.miconda.eu/2015/03/kamailio-2014-awards.html

Looking forward to meeting many of you soon in Berlin, during May 18-20, 2016, at the 4th edition of Kamailio World Conference & Exhibition, to celebrate 15 years of development for Kamailio Project.
Note: I am solely selecting the winners, with no involvement of Kamailio project members, based on what I observed and has risen my interest during 2015.  Also, a rule that I try to enforce is that a winner of a category in the past will not be awarded again same category (a winner one time is a winner for ever).

Enhanced. Fixing. Solving. Enterprise grade. Improving. Completing.

I’ve been seeing this too much lately.

Companies decide to market their product as a way to “fix” WebRTC. The gall.

I understand where this comes from. Marketing is a lot about FUD. How to put fear in your potential customer until the only thing left for him to do is buy.

If you look closely, though, none of them really “fixes” WebRTC. The only thing they are doing is using WebRTC in a way that may fit you as a customer.

An example?

Companies who “fix” WebRTC by adding signalling to it. Or adding authorization. Or having it connect to PSTN.

This isn’t about “fixing”. This is about supporting a specific scenario or feature in a product – not even related to WebRTC itself.

Others “fix” WebRTC by having it work on IE (forcing a plugin on the user or using Flash). Again, less about WebRTC, and more about the use case.

And you know what? WebRTC doesn’t offer notifications either – I am sure you can go ahead and “fix” WebRTC by adding push notifications to your app on top of WebRTC!

WebRTC is a very powerful building block, but that’s about all it is – a building block. You’ll need to add additional building blocks to create a solution with it, so no – you aren’t fixing it – you are just implementing your use case with it.

Please.

Stop fixing WebRTC. It isn’t broken.

Just focus on solving a real world problem for a real customer and be done with it.

The post Everyone and His Dog is Fixing WebRTC appeared first on BlogGeek.me.

Earlier this month Fippo published a post analyzing Slack’s new WebRTC implementation. He did not have direct access or a team account to do a thorough deep dive – not to mention he is supposed to be taking some off this month. That left many with some open questions? Is there more to the TURN network? […]

The post Is Slack’s WebRTC Really Slacking? (Yoshimasa Iwase) appeared first on webrtcHacks.

You may have noticed our shiny new website, featuring our new 2600Hz brand.

This is the new generation of 2600Hz, and we are so excited to share it with you.

First, let’s talk about the new 2600Hz branding. This new modern symbol represents the transmission of signals and data. It symbolizes the connection from point A to point B, which is what we do every single day. Our goal has always been to provide an elegant and powerful format for communications services. 2600Hz strives to be a leader and innovator in the telecom industry, while connecting and empowering businesses through our software.

Now, let’s talk about the next phase of 2600Hz. The new branding represents the growth we have experienced together and is of our plans moving forward. In the beginning, we put the majority of our focus into distributed architecture. We spent years building up our software with heavy emphasis on back-end engineering. While we continue to do so, this next chapter is all about you – our clients – and your customers.

Throughout the next few months you will see a stronger focus on providing tools to help you be successful. These tools range from tight integrations with other services to better developer documentation, allowing people to easily build on the platform. Our goal is to equip you with an arsenal to win more deals quickly and expand your current accounts. We believe that the best way to do this it to foster innovation from not only 2600Hz but from the community as well.

We hope you are as excited as we are! Take a look at our new website and services. There are more exciting announcements coming soon, so stay tuned!

Continuing the program from last years, based on the roots and the tight relation of Kamailio project with the academic environment, we are offering three seats at Kamailio World Conference, May 18-20, 2016, in Berlin, to students enrolled in universities or research institutes (both bachelor and PhD programs qualify). Last year we had students participating from Austria, Slovakia and Netherlands.If you are a student and want to participate, write an email to registration@kamailio.org . Participation to all the content of the event (workshops, conference and social event) is free, but you will have to take care of expenses for traveling and accommodation. Write a short description about your interest in real time communications and what is the university or the research institute you are affiliate to.Also, if you are not a student, but you are in touch with some or have access to students forums/mailing lists, it will be very appreciated if you forward these details.More information about Kamailio World is available on the web site:

• http://www.kamailioworld.com

Looking forward to meeting many of you at Kamailio World 2016!

The biggest news this week is the edition of a Raspberry Pi installation script adding by Ken Rice. Grab yourself a Raspberry Pi and go try it out!

Join us Wednesdays at 12:00 CT for some more FreeSWITCH fun! This week we are talking about Raspberry Pi and the Internet of Things! And, head over to freeswitch.com to learn more about FreeSWITCH support.

New features that were added:

• FS-8932 [mod_managed] Add in process load in-process plugins
• FS-8933 [Debian] Basic FreeSWITCH from source installer that works on Raspbian and Debian. Also installs VertoCommunicator and LetsEncrypt SSL Certs. LetsEncrypt requires the machine to have a public IP and DNS for the FDQN functioning properly in public DNS
• FS-8870 [core] Add human-readable call quality statistics logs on call hangup
• FS-7800 [verto communicator] Added support for calling extra screens with same extension as the original and place the parameter conferenceCanvasID with the desired canvas id into the call parameters in the same place bandwidth preferences are added

Improvements in build system, cross platform support, and packaging:

• FS-8948 [Debian] Handle non-existent configuration in Debian postinst causing failure to copy vanilla configuration on installation
• FS-8841 [Debian] Prevent hanging by pushing events to another queue to be processed in a different sofia thread.
• FS-8942 [build] Pass compiler to libvpx configure to account for environments without gcc
• FS-8952 [Windows] Fixed an issue with unreachable code preventing build and simplify conditions

The following bugs were squashed:

• FS-8938 [mod_conference] Clear res_id when setting the same res_id to another member
• FS-8951 [mod_conference] Video lockup in conference due to race condition
• FS-8957 [mod_conference] Fix for video image blipping on personal canvas mode when 1 participant is watching video on hold
• FS-8943 [configure] Fixed misspellings in two comments
• FS-8950 [mod_skinny] Fixed a few memory leaks
• FS-8946 [mod_xml_cdr] Fixed a segfault on call after loading with no config file or event bind failure causing the module load to fail
• FS-8937 [mod_easyroute] Handle a segfault when using bad customer query or on query error
• FS-8855 [core] Fixed the calculation of variance of tone’s frequency estimator to prevent flawed audio
• FS-8928 [core] Flag a binding error when using EventConsumer::bind with invalid event name instead of blindly using custom
• FS-8168 [core] Use copy image functions from libyuv instead of our home rolled versions as the libyuv versions have optimizations
• FS-8406 [mod_rtmp] Add options to scale down cavas size, fps, and bandwidth
• FS-7915 [mod_sofia] Parse and store multiple path fields
• FS-8663 [mod_sofia] Added saftey checks for ;fs_path= command
• FS-8945 [verto_communicator] Don’t show preview_settings window during a video conference call or you will lose conference video stream until you refresh the page