News from Industry
ClueCon 2016
Summary of WWAN cards configuration
In this github repo, I put together my knowledge about WWAN cards setup, alongside with all initialization scripts.
Filed under: Networking Tagged: 3G, GSM, linux, pcengines, UMTS
Let’s Encrypt – how get to free SSL for WebRTC
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.
FreeSWITCH Week in Review (Master Branch) July 23rd – July 30th
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
Surprise: Free Video Calling is no Guarantee for Success (or Adoption)
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:
- 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
- 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
- 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
- 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
- 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.
FreeSWITCH Week in Review (Master Branch) July 16th – July 23rd
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
FreeSWITCH Week in Review (Master Branch) July 9th – July 16th
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
IETF96 in Berlin
FreeSWITCH Week in Review (Master Branch) July 2nd – July 9th
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 v4.2.8 Released
# cd kamailio
# git checkout -b 4.2 origin/4.2Binaries and packages will be uploaded at:Modules’ documentation:What is new in 4.2.x release series is summarized in the announcement of v4.2.0: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: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!
FreeSWITCH Week in Review (Master Branch) June 25th – July 2nd
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 LTE modem
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 EOTFiled under: Networking Tagged: 3G, GSM, pcengines
Kamailio v4.3.6 Released
# cd kamailio
# git checkout -b 4.3 origin/4.3Binaries and packages will be uploaded at:Modules’ documentation:What is new in 4.3.x release series is summarized in the announcement of v4.3.0: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:Thanks for flying Kamailio and enjoy the summer holidays!
Resetting GSM modules on Yeastar gateways using Ansible
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=8022If 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 v4.4.2 Released
# cd kamailio
# git checkout -b 4.4 origin/4.4Binaries and packages will be uploaded at:Modules’ documentation:What is new in 4.4.x release series is summarized in the announcement of v4.4.0:Thanks for flying Kamailio!
FreeSWITCH Week in Review (Master Branch) June 18th – June 25th
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
FreeSWITCH Week in Review (Master Branch) June 11th – June 18th
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.
Pages
Using the greatness of Parallax
Phosfluorescently utilize future-proof scenarios whereas timely leadership skills. Seamlessly administrate maintainable quality vectors whereas proactive mindshare.
Dramatically plagiarize visionary internal or "organic" sources via process-centric. Compellingly exploit worldwide communities for high standards in growth strategies.
Wow, this most certainly is a great a theme.
Company name
Yet more available pages
Responsive grid
Donec sed odio dui. Nulla vitae elit libero, a pharetra augue. Nullam id dolor id nibh ultricies vehicula ut id elit. Integer posuere erat a ante venenatis dapibus posuere velit aliquet.
Typography
Donec sed odio dui. Nulla vitae elit libero, a pharetra augue. Nullam id dolor id nibh ultricies vehicula ut id elit. Integer posuere erat a ante venenatis dapibus posuere velit aliquet.