News from Industry

Way back in 47 (version that is), Chrome started to mandate the use of HTTPS in conjunction with getUserMedia. To use HTTPS you need a SSL/TLS certificate.  Xander Dumaine covered this a bit for us before, but I still see a lot of people out there struggle with it. As it so happens, the certificate for my […]

The post Let’s Encrypt – how get to free SSL for WebRTC appeared first on webrtcHacks.

This week the ability to add and remove video on re-invites was added.

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-9154 [libsofia] Add & remove video on re-invites

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

• FS-9386 [mod_snmp] Use net-snmp-config for SNMP libs if available
• FS-9385 [mod_conference] Check for ghosts before destroying a conference

The following bugs were squashed:

• FS-9357 [core] Handle packet loss and reset decoder on memory error
• FS-9381 [mod_sofia] Fixed a leak in sofia_presence_chat_send
• FS-9382 [core] Fixed an issue with video broken between two users in verto
• FS-9390 [core] Fixed a ‘Segmentation fault’ during call setup
• FS-9369 [core_media] Added the variable add_ice_candidates=true to enable inserting ice candidates in outgoing sdp
• FS-9394 [mod_av] Fixed the h263 leak

Guess what? Mozilla is removing Hello from Firefox.

It will still be available as an add-on, but it seems to have degraded in its importance to Mozilla, which is understandable.

Goodbye HelloWhat is/was Hello?

Hello was Mozilla’s attempt to build a video calling service. Something that is baked right into the browser, but can be used by any browser supporting WebRTC. Think FaceTime or Hangouts but without the app or even a website.

Mozilla partnered for Hello with TokBox (a Telefonica company), which provided the backend to the service – mainly NAT traversal as far as I can tell.

When Hello was announced, I had my doubts and questions about it.

What went wrong?

A few things were wrong from the onset in Firefox Hello:

1. While it debuted on a desktop browser, its main purpose was mobile. The problem is that Firefox OS got scrapped/pivoted, leaving Hello with no real use
2. It came at a low point in Mozilla’s history. Mozilla partnered during 2014 with 3 vendors, trying to reduce Google’s hold on it: Yahoo, Cisco and Telefonica
   • Yahoo is all but dead – it just got acquired by Verizon
   • Telefonica needed Firefox OS on mobile, and now that that hasn’t matured, my guess is that its interests lie elsewhere these days, so having Telefonica/TokBox as part of Hello probably isn’t helping too much today
   • Cisco only wanted to protect its H.264 investments, which it succeeded
   • This cost Mozilla in focus and diluted its brand from being a pure open alternative
3. Firefox has no real network effect or user base to rely on. It doesn’t connect users to one another but rather it connects viewers to web pages. Having hundreds of millions of viewers doesn’t equate monthly active users for a personal communication tool that is baked into the same product
4. Hello was simple, but offered nothing interesting/innovative/new/needed. People who used apps continued to use apps. Those that wanted to meet over URLs used URLs. Having the button in the browser wasn’t enough to make people leap for the opportunity to use it
5. While available in all WebRTC supporting browsers (=Chrome & Firefox), it was really a Firefox thing. This limited the user base, and especially the ability to start or to really receive a call over a mobile device

The main issue though is that a free video calling service isn’t that much of a deal these days (if this surprises you – just ask Google).

So Mozilla started by embedding Hello right into the browser. Then making it into a system add-on. And now it is making it into just another add-on. I assume it has a lot to do with the usage they’ve seen over the past year for Hello (and its non-adoption). It makes no sense to continue investing the time and effort in it if no one is using it – and having it officially released with the browser once every few months is a waste. Better throw it out of the browser and simplify the browser releases.

The next step might be to sunset the add-on/service altogether and say goodbye to Hello.

Is this predictive to Google’s Duo app?

Google announced Duo and is about to release it. Simplifying things a bit (and dumbing it down), Duo is a FaceTime clone. I covered Allo/Duo a few months back.

On face value, there’s no reason why Google Duo won’t meet a similar fate as Mozilla Hello.

That said, there are a few notable differences:

• Duo is a mobile only app, whereas Hello focused on desktop browsers
• Duo will probably be released on Android and iOS, covering 100% of the mobile market from day one
• Google has a large users base on Android and the ability to get Duo in front of users. It also has the social graph of these people – via the phone’s address book
• While Google kept Duo simple, it did bake two features into it:
  • Speed of connectivity, taking it to the extreme by adding QUIC into the mix
  • Caller’s video sent even before you accept the call

Will this be enough for Google Duo to get the adoption? I don’t know.

Where do we go from here?

In 2016 there should be no doubt anymore:

If you plan to monetize a video calling service, you need a serious business plan.

Most services I see launched have no business plan. They attempt to grow to millions of users. There’s a lot of dumb luck involved in it.

I’ve had my doubts about the viability of Wire as a company due to the same reasons. The only progress made by Wire is open sourcing their app – this doesn’t strike me like a business plan or a signal of strength and healthy growth.

 

Planning on introducing WebRTC to your existing service? Schedule your free strategy session with me now.

The post Surprise: Free Video Calling is no Guarantee for Success (or Adoption) appeared first on BlogGeek.me.

This week we saw the addition of customized video mute banners in mod_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-9230 [mod_conference] Customize video muted banner

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

• FS-9373 [Debian] Added mod-verto and mod-rtc to freeswitch-meta-all package

The following bugs were squashed:

• FS-9355 [core] Fixed a segfault in case of null frame
• FS-9356 [core] Fixed an issue with DTMF not recognized when coming from a Cisco SIP trunk
• FS-9353 [mod_conference] Fixed a problem with clear-vid-floor producing an error while working
• FS-9259 [mod_spandsp] Fixed a missing “m=image 0” when replying to INVITE with disable image line
• FS-9289 [core] Fixed a MOH issue with b side hold causing silence for the a leg
• FS-9365 [core] Fixed the SDP format on reply to RE-INVITE to be RFC-4566 compliant
• FS-9357 [verto communicator] Fixed an issue with VP9 codec screensharing on mod_conference (mux/transcode) not working
• FS-9342 [verto_communicator] Fixed a problem with settings not being saved when closing the settings panel
• FS-9368 [mod_sofia] Fixed a problem with errant duplicate video frames causing video recording issues
• FS-8783 [libsrtp] Fix alignment issue
• FS-9376 [mod_sofia] Fixed a hold negotiation problem on a call received from a Cisco Session Manager

This was a quiet week with a few bug fixes and a minor configuration update.

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

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

• FS-9350 [configuration] Add mod_av commented to modules.conf.xml

The following bugs were squashed:

• FS-9342 [verto_communicator] Properly saving settings to localStorage when closing the settings panel
• FS-9345 [mod_httapi] Fixed an issue with HTTAPI truncating a string when responses span multiple packets
• FS-9343 [mod_smpp] Fixed a problem with failing to send a message via Nexmo
• FS-9352 [core] Fixed overzealous ptime adjust issues on opus

The 96th meeting of IETF (the Internet Engineering Task Force) takes place in Berlin, Germany, during July 17-22, 2016. Ahackaton tied to the IETF meeting is organized during the weekend, July 16-17 .Among the major standardization topics to be discussed, from Kamailio point of view, are: SIP, WebRTC, TURN, IPv6, TLS and DNS (DNSEC/DANE).Daniel-Constantin Mierla and Olle E. Johansson from the Kamailio community will be present at the event.Together with Lorenzo Miniero from Janus WebRTC Gateway project, they plan to organize a meetup (for drinks/dinner) on Monday evening, July 18 — likely to happen at a restaurant nearby IETF meeting, starting around 20:00. If you are in Berlin and want to join, contact us (email to sr-users mailing list or contact directly one of these three persons). Each participant will take care of own expenses, we aim to have an open discussion about what’s new lately and where we head on in real time communications space.Looking forward to meeting some of you next week in Berlin!Thanks for flying Kamailio!

This week we have three great features to announce! First, the addition of mod_sms_flowroute! Second, amplitude estimation in mod_avmd. This particular addition will be neat for those math enthusiasts out there. And finally, mod_dptools got two new API calls.

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-9009 [mod_avmd] Amplitude estimation
• FS-9310 [mod_sms_flowroute] Added native support for Flowroute SMS API over HTTP(S)
• FS-9264  [mod_dptools] Add detect_audio and detect_silence API calls

The following bugs were squashed:

• FS-9241 [mod_sofia] Use tls_public_url instead of tls_url in INVITE Contact when NAT is detected
• FS-9316 [mod_sofia] Fixed an issue caused by INVITE with empty SDP from Cisco VCS not setting up video
• FS-9328 [core] Fixed switch_jb_peek_frame bug where it uses the len of the whole packet and does not subtract the len of the rtp header when copying and returning the size of the packet read.
• FS-9333 [mod_sofia] Disable video refresh by sip INFO by default because this method is outdated
• FS-9337 [core] Fixed invalid sdp generated with soa disabled

Kamailio SIP Server v4.2.8 stable is out! This is a minor release including fixes in code and documentation since v4.2.7.Kamailio v4.2.8 is based on the latest version of GIT branch 4.2.  If you are running previous 4.2.x versions are advised to upgrade to 4.2.8 (or to 4.3.x/4.4.x series). If you upgrade from older 4.2.x to 4.2.8, there is no change that has to be done to configuration file or database structure comparing with older v4.2.x.Resources for Kamailio version 4.2.8Source tarballs are available at:

• http://www.kamailio.org/pub/kamailio/4.2.8/src/

Detailed changelog:

• http://www.kamailio.org/pub/kamailio/4.2.8/ChangeLog

Download via GIT: # git clone git://git.kamailio.org/kamailio kamailio
# cd kamailio
# git checkout -b 4.2 origin/4.2Binaries and packages will be uploaded at:

• http://www.kamailio.org/pub/kamailio/4.2.8/

Modules’ documentation:

• http://www.kamailio.org/docs/modules/4.2.x/

What is new in 4.2.x release series is summarized in the announcement of v4.2.0:

• http://www.kamailio.org/w/kamailio-v4-2-0-release-notes/

Note: the branch 4.2 is an old stable branch. The latest stable branch is 4.4, at this time with v4.4.2 being released out of it. The project is officially maintaining the last two stable branches, these are 4.4 and 4.3. Therefore an alternative is to upgrade to latest 4.4.x – be aware that you may need to change the configuration files and database structures from 4.2.x to 4.3.x/4.4.x. See more details about them at:

• http://www.kamailio.org/w/kamailio-v4-3-0-release-notes/
• http://www.kamailio.org/w/kamailio-v4-4-0-release-notes/

Important: this version marks the end of planned releases from branch 4.2. From now on the focus is on maintaining the branches 4.4 and 4.3 for stable releases.Thank you for flying Kamailio! Enjoy the summer holidays!

This week was filled with bug fixes and build improvements. This week also marks the one month mark until ClueCon, so be sure to sign up and book a hotel room so you don’t miss out!

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

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

• FS-9317 [configuration] Added screen share examples to the vanilla configurations
• FS-9320 [mod_local_stream] When the entity playing the local_stream video file has a greater or equal frame rate, reduce the buffering
• FS-9315 [mod_http_cache] Added support for video file formats

The following bugs were squashed:

• FS-9301 [mod_sofia] Handled a race condition on startup of mod_sofia with error conditons causing segfault
• FS-9302 [mod_mongo] Fixed mongo_find_one and mongo_find_n to return -ERR when the connection to the database fails
• FS-9221 [mod_conference] Add inactive support for calls to prevent termination if just the video stream is removed
• FS-9303 [mod_conference] Removed unnecessary checks as the video flag is not sent to file open unless using transcode mode, you can record mp4 but it will only contain the audio if in passthru mode
• FS-9305 [mod_conference] Fix for fs_cli crashing due to vid-logo-img incorrectly being set to nothing after originally setting it to a bad image
• FS-9307 [mod_conference] Fixed a race condition caused by trying to use a closed file handle when playing a video file after closing files before video threads are done
• FS-9313 [mod_opus] Fixed sprop_stereo interpretation causing bad audio
• FS-9312 [core] Fixed and unreachable code block in switch_core_media
• FS-9314 [mod_conference] Fixed a crash when starting conference in mux mode while specifying or defaulting to a layout group that does not exist. We will now fall back to transcode mode in this case

Huawei ME909s-120 is the newest modem of Huawei LTE/UMTS family, and it is sold for around $70 at TechShip.se and at Aliexpress.

The modem is immediately recognized as CDC Ethernet device in Debian 8 kernel, and is visible as usb0 interface. In the scripts below, the ttyUSBx serial ports are aliased to ttyWWANxx, and usb0 is renamed to lte0, in order to avoid any naming conflicts with other devices, and also to avoid possible name changes  due to a kernel upgrade.

cat >/etc/udev/rules.d/99-huawei-wwan.rules <<'EOT' SUBSYSTEM=="tty", ATTRS{idVendor}=="12d1", ATTRS{idProduct}=="15c1", SYMLINK+="ttyWWAN%E{ID_USB_INTERFACE_NUM}" SUBSYSTEM=="net", ATTRS{idVendor}=="12d1", ATTRS{idProduct}=="15c1", NAME="lte0" EOT cat >/etc/chatscripts/sunrise.HUAWEI <<'EOT' ABORT BUSY ABORT 'NO CARRIER' ABORT ERROR TIMEOUT 10 '' ATZ OK 'AT+CFUN=1' OK 'AT+CMEE=1' OK 'AT\^NDISDUP=1,1,"internet"' OK EOT cat >/etc/chatscripts/gsm_off.HUAWEI <<'EOT' ABORT ERROR TIMEOUT 5 '' AT+CFUN=0 OK EOT cat >/etc/network/interfaces.d/lte0 <<'EOT' allow-hotplug lte0 iface lte0 inet dhcp     pre-up /usr/sbin/chat -v -f /etc/chatscripts/sunrise.HUAWEI >/dev/ttyWWAN02 </dev/ttyWWAN02     post-down /usr/sbin/chat -v -f /etc/chatscripts/gsm_off.HUAWEI >/dev/ttyWWAN02 </dev/ttyWWAN02 EOT
Filed under: Networking Tagged: 3G, GSM, pcengines

Kamailio SIP Server v4.3.6 stable is out – a minor release including fixes in code and documentation since v4.3.5. The configuration file and database schema compatibility is preserved.Kamailio (former OpenSER) v4.3.6 is based on the latest version of GIT branch 4.3, therefore those running previous 4.3.x versions are advised to upgrade. There is no change that has to be done to configuration file or database structure comparing with older v4.3.x.Resources for Kamailio version 4.3.6Source tarballs are available at:

• http://www.kamailio.org/pub/kamailio/4.3.6/src/

Detailed changelog:

• http://www.kamailio.org/pub/kamailio/4.3.6/ChangeLog

Download via GIT: # git clone git://git.kamailio.org/kamailio kamailio
# cd kamailio
# git checkout -b 4.3 origin/4.3Binaries and packages will be uploaded at:

• http://www.kamailio.org/pub/kamailio/4.3.6/

Modules’ documentation:

• http://www.kamailio.org/docs/modules/4.3.x/

What is new in 4.3.x release series is summarized in the announcement of v4.3.0:

• http://www.kamailio.org/w/kamailio-v4-3-0-release-notes/

Note: the branch 4.3 is the previous stable branch. The latest stable branch is 4.4, at this time with v4.4.2 being released out of it. The project is officially maintaining the last two stable branches, these are 4.4 and 4.3. Therefore an alternative is to upgrade to latest 4.4.x – be aware that you may need to change the configuration files and database structures from 4.3.x to 4.4.x. See more details about it at:

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

Thanks for flying Kamailio and enjoy the summer holidays!

Sometimes there’s a need to reset a GSM module on a Yeastar GSM gateway. For example, SIM cards of one of our providers get into faulty state every few weeks, and only a reset helps.

The GSM module can either be rebooted via Web GUI, or from the Asterisk console. But the Asterisk console can only work on the same host where the asterisk daemon runs, so you need to make an SSH connection into the Yeastar box to do that. Also it’s impossible to save a public SSH key in a Yeastar box, so only password authentication works.

Ansible is a powerful toolset for managing remote hosts, and it appears to be perfectly suitable for managing the GSM gateways.

Ansible 2.x is available for Debian 8 from jessie-backports repository. There are some important differences from version 1.7 that is installed from default repositories, and in particular, ansible_host and ansible_port variables.

After installing Ansible, uncomment host_key_checking = False in /etc/ansible/ansible.cfg , so that the SSH client stops verifying the remote host SSH signatures. Otherwise the host signatures should be listed in your known_hosts file.

The following lines in /etc/ansible/hosts list your GSM gateways:

[yeastar] gsm01 ansible_host=192.168.99.66 ansible_ssh_pass=kljckhjeswvdfesv gsm02 ansible_host=192.168.99.67 ansible_ssh_pass=dmnckjfvrever gsm03 ansible_host=192.168.99.68 ansible_ssh_pass=dcmnkljdfhfe [yeastar:vars] ansible_user=root ansible_port=8022

If you use the same root password on all devices, the password variable can be moved to the group variables.

Ansible uses SFTP for ad-hoc commands, and SFTP is not available on Yestar gateways. But the raw module works just fine, and resetting a GSM module can now be done with a simple command from your management server:

ansible gsm03 -m raw -a '/bin/asterisk -rx "gsm power reset 2"'

 

Filed under: Networking Tagged: GSM, linux, pbx, sip, voip

Kamailio SIP Server v4.4.2 stable is out – a minor release including fixes in code and documentation since v4.4.1. The configuration file and database schema compatibility is preserved.Kamailio v4.4.2 is based on the latest version of GIT branch 4.4, therefore those running previous 4.4.x versions are advised to upgrade. There is no change that has to be done to configuration file or database structure comparing with older v4.4.x.Resources for Kamailio version 4.4.2Source tarballs are available at:

• http://www.kamailio.org/pub/kamailio/4.4.2/src/

Detailed changelog:

• http://www.kamailio.org/pub/kamailio/4.4.2/ChangeLog

Download via GIT: # git clone git://git.kamailio.org/kamailio kamailio
# cd kamailio
# git checkout -b 4.4 origin/4.4Binaries and packages will be uploaded at:

• http://www.kamailio.org/pub/kamailio/4.4.2/

Modules’ documentation:

• http://www.kamailio.org/docs/modules/4.4.x/

What is new in 4.4.x release series is summarized in the announcement of v4.4.0:

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

Thanks for flying Kamailio!

This week there was an improvement to mod_spandsp with a channel variable added to make spandsp_start_tone_detect easier to use from the dialplan or embedded scripts

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-9287 [mod_spandsp] Add channel variable to make spandsp_start_tone_detect easier to use from dialplan/embedded scripts.

The following bugs were squashed:

• FS-9283 [mod_hiredis] Fixed an issue with using hiredis_raw on channels without media such as an originate
• FS-9292 [core] Fixed a core dump while playing videos or showing images usually with a high number of callers
• FS-9296 [mod_httapi] Fixed video support
• FS-9297 [mod_sofia] Fixed multiple crashes from passing invalid null values in sofia.conf

This week we had a new feature in mod_sofia. A new parameter was added, renegotiate-codec-on-hold, for proxy hold when proxy media and proxy mode are disabled; its similar to proxy-refer.

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-9192 [mod_sofia] Added renegotiate-codec-on-hold parameter for proxy hold when proxy media and proxy mode are disabled; its similar to proxy-refer

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

• FS-9263 [build] Attempting to find the proper lua5.2 version on openbsd
• FS-9260 [build] Fixed make detection to not fail on openbsed,  fixed libtoolize detection to attempt to find libtoolize the same version as specified libtool, and added -ltermcap for openbsd so it can correctly link to libedit

The following bugs were squashed:

• FS-9244 [core] Fixed debug lines
• FS-9265 [core] Fixed an issue with receiving INCOMPATIBLE_DESTINATION when there is no RTCP
• FS-9271 [mod_conference] Fixed a segfault trying to record a canvas that does not exist
• FS-9267 [mod_cv] Fixed an issue where the VPX codec returns the same image to the core when doing repeated decoding. Updates to that image match the updates to the stream so if a media bug modifies the image between key frames it messes up the picture until the next key frame is received.

Hardware acceleration for video codecs is almost mandatory.

VP9 is getting a performance boost

There are three things that keep VP8 in the game when compared to H.264:

1. It was the only video codec in Chrome for WebRTC in the last 5 years, giving it a headstart in deployments
2. H.264 while available in mobile chipsets isn’t always accessible for the developer (or works as it should when it is accessible)
3. VP8 and H.264 are rather old now, so software implementations of them are quite decent

 

With VP9, the main worry was that it will be left behind and not get the love and attention from chipset vendors – leading it to the same fate as VP8 – abysmal, if any, hardware acceleration support. It is probably why Google went to great lengths to make it running on YouTube so soon and is publicizing its stats all the time.

This worry is now rather behind us. Recent signs show some serious adoption from the companies that we should really care about:

#1 – ARM

Mobile=ARM

Without checking stats, I’d say that 99% or more of all smartphones sold in the past 5 years are based on ARM.

If and when ARM decides to support a feature directly, that brings said feature very close towards world domination in future smartpones.

Which is somewhat what happened last week – ARM announced its Mali Egil Video Processor with VP9 acceleration.

Here’s a deck they shared:

ARM Mali "Egil" technical preview from Phil Hughes

Being farther away from chipsets than I were 5 years ago, it is hard for me to say if this is an integral part of an ARM processor, but I believe that it isn’t. It is an add-on component that takes care of video processing that chipset vendors add next to their ARM core. They can source the design from ARM or other suppliers – or they can develop their own.

Not sure how popular the ARM alternative is for video processing, but they have the advantage of being the first alternative for any chipset vendor (hell – they already source the ARM core itself, so why not bundle?). Which also means every other vendor needs to match up to their feature set – and improve on it.

Now that VP9 encode/decode capabilities are front and center in the ARM Mali Egil, it has become a mandatory checkmark for everyone else as well.

#2 – Intel

If ARM is the king of mobile, then Intel rules the desktop.

As with ARM, I haven’t been following up on Intel CPU acceleration lately. And as with ARM, it was Fippo who got my attention with this link here: the new Intel Media SDK.

For those who don’t know, Intel is providing several interesting software packages that make direct use of its chipset capabilities. Especially when it comes to optimizing different types of workloads. The Intel IPP and Media SDKs handle media related processing, and are quite popular by low level developers who need access to such facilities.

From the release page itself:

With this release we are happy to announce new full hardware accelerated support for HEVC and VP9.

• HEVC Main 10 (10-bit) encoder and decoder support
• VP9 8-bit and 10-bit decoder support

So… HEVC (=H.265) has encode and decode while VP9 only has decode support.

Probably because HEVC has been in the works for a lot longer than VP9, but there’s hope still.

#3 – Alliance of Open Media

The Alliance of Open Media. I’ve published a recent update on the alliance.

Intel was there from the start. The recent additions include ARM, AMD and NVIDIA.

I am sure additional chipset vendors will be joining in the coming months – there seems to be a ramp up in memerships there, with Ateme and Adobe added to their logos just last week.

While the alliance is about what comes after VP9, it is easy to see how these vendors may sway to using VP9 in the interim.

The Future

The future is most definitely one of royalty free video codecs. We’ve got there with voice, now that we have OPUS (though Speex and SILK were there before to pave the way). We will get there with video as well.

Coding technologies need to be accessible and available to everyone – freely – if we are to achieve Benedict Evans’ latest claims: Video is the new HTML. But for that, I’ll need another post.

 

Planning on introducing WebRTC to your existing service? Schedule your free strategy session with me now.

The post VP9 Hardware Acceleration is Real appeared first on BlogGeek.me.

Our good old Samsung Galaxy Tab 3 7.0 Kids Tablet has finally died after over 3 years of everyday heavy use, so I needed a new solution. So far, here is the best combination that I could find:

This silicon case for Samsung Galaxy TAB A 7″ SM-T280 is a solid and protective piece, and it allows the kids hold the tablet with their little hands without slipping off. It also works as a stand, so it’s very convenient for watching videos.

The Samsung Galaxy Tab A (7″, 8GB, Metallic Black) fits perfectly into the protective case. The tablet is coming with preinstalled “Kids Mode” application, which is pretty neat, but very restrictive: the kid can watch only the videos on SD card that you mark as safe, and YouTube is not available. You can install kid-safe YouTube wrappers from the Play market, but it’s a bit too much hassle to my taste.

So, instead of the Samsung Kids Mode, I installed Kids Place by kiddoware. With a little payment, you get a good child protection mode, and you can enable YouTube directly on the child screen. The payment is also transferable to other devices under your account.

Also, this portable Bluetooth speaker works as a stand for a tablet, and it produces a much better sound than the tablet’s own speaker. Unfortunately the silicon case is too thick for this stand, but it’s a minor issue, and the speaker can easily be placed behind the tablet.

 

Filed under: Hardware Tagged: kids