- WebRTC gateway 1.77.16-514 is available on ALE Business Portal since September 17.
1.77.16-514
- Bug fixes:
- new WRTRANGE and SIPRANGE ports range options for mpconfig.
mpconfig --WRTRANGE=20000-29999 for WebRTC ports range.
mpconfig --SIPRANGE=30000-39999 for SIP ports range.
- add VirtIO disk block device support.
- mpcheck DNS tests reworked.
- mpcheck test only relevant port on media via proxy.
- set default log level to info instead of verbose.
- reformat janus MOS audio ticket.
- Bug fixes:
- Fixed no audio issue when the proxy is also used for media over proxy (mpmproxy = on).
- Fixed call establishment issues (occurred after a crash of janus, solved with janus version 0.10.10).
- WebRTC gateway on Hyper-V is now officially supported, see the README.txt file in the WebRTC gateway zip package for installation procedure.
see previous change logs
1 Installation
1.1 Requirements:
- Rainbow users need to be granted with either a Business or an Enterprise licenses and a mandatory Enterprise License when connected to a 3rd Party PBX.
- OmniPCX Enterprise PBX needs:
- to run with SW version OmniPCX Enterprise from R12.1 MD2 (M2.300.21.a) & OmniPCX Enterprise R12.2 (M3.402.13.f).
- to be granted with REX licenses per targeted Rainbow user
- to be connected to Rainbow for the Telephony Services
- A Virtual Machine must be set-up to host Rainbow WebRTC Gateway. The VM software is delivered by ALE and requires a VMWARE 6.0 or later or on Hyper-V. The VM has to meet the following minimum requirements that may still change:
- vCPU: 2
- CPU reservation in MHz: 4800
- vRAM: 2GB
- vDISK: 4 GB
- vNIC: 1 GB
- OXO Connect PBX needs:
- to run with latest OXO Connect SW version (at the date of publication of this article R310/017.001)
- or to run at least R400/507.002 to use an OCE Front End as WebRTC gateway
- to be granted with an AnyDevice Terminal per targeted Rainbow user
- to be connected to Rainbow for the Telephony Services
- to have VoIP Channels for the SIP trunk
- For OXO Connect you can also host the Rainbow WebRTC Gateway on a standalone PC. Read the dedicated installation guide for more details. Or on an OCE WebRTC Front End (OCE FE) which is an OCE hardware running only the WebRTC Gateway component. Read the dedicated article for OCE FE configuration through Rainbow.
Details about the the Rainbow offer connected to a OmniPCX Enterprise or OXO Connect/OXO Connect Evolution can be found in following Technical Business Engine presentation.
- 3rd party PBX needs:
- to run with with latest version of Rainbow CTI & Media bridge
- to be connected to Rainbow for the Telephony Services by the the CTI & Media bridge
- to have a SIP trunk connected with the CTI & Media bridge
- See Section CTI & Media bridge for other requirements and supported PBXs
1.2 WebRTC gateway OVF/ISO installation:
1.2.1 Software Download
1.2.2 Installation:
First deploy the OVF file in your vCenter and start the virtual machine (VM) or if connected to an OXO Connect install the ISO file on a standalone PC (installation guide).
Note: The OVF and ISO have been created with a QWERTY keyboard, you may need to change it:
Login with following credentials:
User=kb
Password=kb
The wizard to change the keyboard will be automatically started, just follow the instructions on the screen to change your keyboard.
You may also access to the same menu by using following commands:
# dpkg-reconfigure keyboard-configuration
# service keyboard-setup restart
For the configuration login with following credentials on the console:
User=rainbow
Password=Rainbow123
For root commands you must use “sudo”
1.3 WebRTC Gateway configuration
When logged using rainbow account, the configuration is made using the commands mpnetwork [options...] and mpconfig [options...]
mphelp //displays the list of mpXXXXX commands//
mpstart|mpstop|mprestart //start, stop or restart gateway//
mpcheck //checks WebRTC gateway connectivity to Rainbow Cloud//
1.3.1 First Installation:
You need first to configure the Network settings then the WebRTC gateway settings.
By default the Rainbow WebRTC Gateway starts using a DHCP based network setting (DHCP has only to be used with OXO Connect Zero Touch).
Login with the rainbow account to make the configuration.
1.3.1.1 Step 1 - Network configuration
Consult first What Are Rainbow Network Requirements?
The 1st time you have to run the command mpnetwork with the mandatory options IP, NETMASK, GATEWAY and DNS as shown below:
mpnetwork --IP="10.2.3.4" --NETMASK="255.255.255.0" --GATEWAY="10.2.3.254" --DNS="10.2.3.253"
You can also run for this 1st step the mpnetwork command with all options by adding NTP and HOSTNAME
mpnetwork --IP="10.2.3.4" --NETMASK="255.255.255.0" --GATEWAY="10.2.3.254" --DNS="10.2.3.253" --NTP="10.2.3.253" --HOSTNAME="rainbow-mpgw"
optional you can add following options
--HOSTDOMAIN="mycompany.com"
--PROXY="http://user:pass@10.2.3.252:8000/"
or
--PROXY="http://10.2.3.252:8000/"
In case you need to update the configuration each command can be executed separately
mpnetwork --IP="10.2.3.4" // your new ip address of the VM //
mpnetwork --NETMASK="255.255.255.0"
mpnetwork --GATEWAY="10.2.3.254" // your gateway //
mpnetwork --DNS="10.2.3.253" // your DNS //
mpnetwork --NTP="10.2.3.253"
mpnetwork --HOSTNAME="rainbow-mpgw"
mpnetwork --IP=dhcp // if you want to switch IP config to DHCP mode (only with OXO Connect Zero Touch) //
you may also use
mpnetwork --FQDN="rainbow-mpgw.mycompany.com"
1.3.1.2 Step 2 - Configure the WebRTC Gateway:
The 1st time you have to run the command with the mandatory options PBX_DOMAIN and PBXID as shown below:
mpconfig --PBX_DOMAIN="10.2.3.5" --PBXID="PBXa1b1-2wxy-3c3d-6789-4e4f-g556-7h89-10i9"
PBX_DOMAIN= is your OXE/OXO ip address
PBXID= is your OXE/OXO pbxid in Rainbow
With this new option the WebRTC gateway uses 2 TURN servers to establish the calls.
- In case of upgrade we recommend to run following command:
mpconfig --TURN_SERVER=GEOIP
- For a new installation the TURN option is default GEOIP.
optional you can use FQDN instead of IP Addresses
mpconfig --RAINBOW_DOMAIN="domain.yy.zz"
mpconfig --RAINBOW_HOST="host.uu.vv"
- PBX_DOMAIN can be an FQDN (in this case the DNS used by the WebRTC Gateway must resolve this FQDN)
- In case of spatial redundancy from OXE Call servers, the PBX_DOMAIN must contain the node name configured in netadmin, section 17. 'Node configuration'. This FQDN must be resolved by the DNS through a DNS delegation to the OXE system.
mpconfig --TURN_SERVER="turn-abx.openrainbow.com"
mpconfig --TURN_PORT="3478" Choose the TURN server by replacing abx with the value of the TURN server that is the nearest to your location. List of TURN servers can be found in the document attached to the article What Are Rainbow Network Requirements?
mpconfig --xxx
mpapply
mprestart
1.3.1.3 Step 3 - Check configuration
Once the server has restarted you can run the command mpcheck to check the connectivity to Rainbow Cloud.
1.3.1.4 Config files
Once you have executed both scripts mpnetwork and mpconfig you can check the configuration in following file /etc/rainbow-mp.cfg.
IP="10.2.3.4"
NETMASK="255.255.255.0"
GATEWAY="10.2.3.254"
HOSTNAME="rainbow-mpgw"
HOSTDOMAIN="mycompany.com"
DNS="10.2.3.253"
NTP="10.2.3.253"
LOGLEVEL="info"
JANUSLOGLEVEL="4"
PROXY=""
TURN_SERVER="GEOIP"
TURN_PORT="3478"
PBXID="PBXa1b1-2wxy-3c3d-6789-4e4f-g556-7h89-10i9"
PBX_DOMAIN="10.2.3.5"
RAINBOW_DOMAIN="openrainbow.com"
RAINBOW_HOST="openrainbow.com"
This is a summary of the parameters that will be used to configure the VM and the 3 services of the WebRTC gateway.
/etc/janus/janus.cfg
/etc/otlitemediapillargateway/config.js
/etc/kamailio/kamctlrc
1.3.2.3 Change network settings to DHCP (Only with OXO Connect Zero Touch)
run the command:
mpnetwork --IP=dhcp
optional you can add a proxy
mpnetwork --PROXY="user:pass@10.1.2.252:8000/"
After the modification reboot the server with the command sudo reboot
2 PBX Configuration
2.1 OminiPCX Enterprise configuration:
Please consult TC2462.
2.2 OXO Connect / OXO Connect Evolution
Please consult TC2479.
3 Rainbow Configuration:
3.1 Equipement configuration
For the OmniPCX Enterprise or OXO Connect connected to the WebRTC Gateway you'll have to select the option "Activate the WebRTC gateway"
3.2 Users Configuration
Each user needs to be granted with either a Business or an Enterprise license
And the Rainbow account must be associated to his PBX phone
4 Troubleshooting:
for troubleshooting please read the troubleshooting guide
5 Old change logs:
1.76.7-391
- Bug fixes:
- Solved mpcheck issue for TURN server connectivity check.
- Better anonymised logs regarding internal server connection credential.
- Solved some audio issues in case of SIP encryption
- Solved call cuts after some time (more or less 50min); related to Janus crashes that have been corrected.
- Fixed issue on GEOIP check in mpcheck.
- Removed turn_server=turn-eu1.openrainbow.com and replaced by TURNSERVER=GEOIP in shell history.
- Fixed mpcheck issue when proxy is used.
- Fixed issue of log file rotation of syslog which was only writing writing logs into the file syslog.1.
- Solved random lost of VoIP services with implementation of a xmpp ping keepalive mechanism.
- Solved connection issue when FQDN is used for the proxy configuration.
- Fixed Linux security issue CVE-2021-3156.
- janus meetecho base upgraded to 0.10.4.
- add TURN servers TTL check and self test at startup.
- add SIP test to mpstc (module for stun/sip/ntp/Rainbow server reachability test).
- implementation of SIP/TLS SRTP WebRTC gateway connected to OXE.
- add xmpp ping keepalive mechanism.
- Bug fixes:
in case of OXE with spatial redundancy use mpconfig --SSIP=IP_1_OF_PBX,IP_2_OF_PBX
- Bug fixes:
- Fixed issue WebRTC gateway connection KO after upgrade to 1.75.3 when using proxy.
- Fixed issue TURN server IP address is used instead of FQDN when mpmproxy is used for media over proxy.
- Enhancements:
- Fixed issue one way audio when PBX codec is OPUS (OXO Connect Evolution).
- Fixed issues on SRTP (OXE -WebRTC gateway); SIP-TLS signaling and handling of 2 audio media lines in SDP.
- Fixed audio loss and call cut on Android.
- Fixed issue changing the IP address of the WebRTC gateway was not taken into account (SIP domain and configuration file were not updated).
- component upgrade janus to 0.8.2.
- component upgrade libnice to current master 0.1.16.0-3.
- component update nodejs 8 dependencies.
- Enhancements:
- Fixed issue user registration lost after internal error (missing call notification).
- Fixed authentication error message in mpstatus after upgrade to 1.74.10.
- Fixed Network card not recognized on NUC8BEH.
- ensure media proxy cleanup when disabling proxy.
- cleanup portal /tmp files at service start and stop.
- added command mpcleanlog to clean deamon and syslog log files in case logrotate didn't work correctly (issue is that all deamon and syslog logs are written into the files deamon.1 and syslog.1 instead of creating new files as defined by logrotate).
- Bug fixes:
- Fixed call cuts trying to communicate with an Android in bad network coverage environment. The same mechanism of "renegotiation" fixes this issue.
- Fixed issue about missing call notification on mobile client due to process crash and unregistered devices.
- Fixed some audio loss issue during web conference.
- Fixed some mpchek STUN/TURN test issues.
- regular retry to update the turn server list in case of potential failure.
- SIP register sessions are now stored into the WebRTC gateway. Improves VoIP feature recovery in case of WebRTC gateway restart.
- Add VM hypervisor type in case of VM.
- Grub boot workaround when no screen is connected. The state of connected / disconnected screen is available through the command mpcollect.
- Serviceability improvements and bug fixes:
- Some call cuts issues have been fixed.
- mpcheck STUN/TURN tests have been reworked.
- Serviceability improvements and bug fixes.
- Coming soon new feature to avoid ringing a disconnected device.
(for OmniPCX Enterprise only and requires a specific OmniPCX Enterprise version, availability of this version will be announced when released.)
mpconfig --RINGINGAUTO=[true|false].
true (default value): Ringback tone is played when calling a disconnected device.
false: The call will be hung up when a disconnected device is called.
1.73.9-140
- Serviceability improvements and bug fixes.
1.73.4-113
- Geolocalization for TURN servers (mpconfig --TURN_SERVER=GEOIP).
- LOGLEVEL set to verbose at upgrade time.
- Security patch CVE-2019-11477.
- Serviceability improvements and bug fixes.
1.71.4-61
- WebRTC gateway can now be connected to an OXO Connect or OXO Connect evolution that is networked with another PBX.
- Improvements to the option to send media streams through proxy.
- serviceability improvements and bug fixes.
1.70.3-175
- added option to send also media streams through proxy (mpmproxy on|off).
- serviceability improvements and bug fixes.
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