VoIP Calling: WebRTC gateway change log

Below you'll find the complete change log history for the WebRTC gateway since version 1.70.3-175.

2.2.5-516
Bug fixes:

- Fixed lost of audio after a call is transferred/forwarded.

- Fixed issue of user unable to make or receive a call.

- Fixed installation issue on standalone PC.

- Kernel Upgrade.

2.2.1-361
Bug fixes:

- Fixed SIP TLS issue with OXE (Noise at start/during call).

- Fixed some memory leak problems.

- Kernel Upgrade.

Serviceability:

- EFI boot reworked.

- improvements of the logs management.

- and many other enhancements.

2.1.8-302
Important: This release is intended to be used as a patch to resolve the DTMF issue. Do not install this release if you are using SIP encryption between PBX and the WebRTC gateway. Due to this restriction, this release will not be available for remote update and must be downloaded from MyPortal.
Bug fixes:

- Fixed non working DTMF during a call on OXE and OCO/OCE.
- Reconnection mechanism improved.

- Serviceability :

- EFI boot reworked.
- Logs management improved.
- mpsystem, mpcheck commands reworked and enhanced.

2.1.0-215
Bug fixes:

- Fixed problem of not sending DTMF in WebRTC gateway context with OXE.

- Improved support of large systems (more then 1500 users).

- In case of Duolan configuration each ethernet port has a dedicated role:

eth1 only for webRTC media and eth0 only for SIP media

2.0.3-204
New feature:

- Support of SDP protocol Unified Plan

- Bug fixes :

- Fixed problems of one way audio
- Fixed issues of call cuts after a few minutes
- Fixed issue of missing call notification on Rainbow Client
- Fixed issue when setting up a Zero Touch configuration with OXO

- Serviceability improvements and enhancements:

- OS upgraded to Debian 11 (bullseye)
- enhanced portal logs for R&D support purposes

1.78.11-470
Bug fixes:

- fixed some certificate management issues for SIP/TLS feature

- Serviceability improvements and enhancements:

- fixed mpcollect
- rework of mpsystem

1.78.6-449
Bug fixes:

- Fixed problem when using Proxy for signaling AND media streams.

1.78.5-447
Bug fixes:

- Fixed problems of one way audio for calls with Android Rainbow app.

1.78.4-444
- Bug fixes:

- Fixed memory leak issue. Main symptoms were inbound/outbound calls were not working, call cuts and no audio during calls.
- Fixed problems of randomly missing audio (both and one way) for incoming and outgoing calls.

- Serviceability improvements and enhancements:

- Added new command mpreboot to reboot the WebRTC gateway
- Added new command mphalt to power off the WebRTC gateway
- Added new command mpsystem to display system status (CPU type/usage, RAM size/free, etc...)

1.77.16-514
- Bug fixes:

- Unable to initiate a call in Computer Mode with Rainbow (network instability).

- Serviceability improvements and enhancements:

- New installer script available for Hyper-V G2.
- 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.

1.77.5-409
- Bug fixes:

- Fixed mpcheck issue when PBX is declared with its FQDN instead of its IP Address.
- 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).

- Serviceability improvements and enhancements:

- janus upgraded to 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.

1.76.7-391
- Bug fixes:

- Solved one-way audio for calls using G722 codec.
- 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.

- Serviceability improvements and enhancements:

- libnice upgraded to 0.1.17.
- 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.

1.75.6-311
- Bug fixes:

- Fixed an issue that allowed the registration of devices whose IP address did not match that of the PBX Equipment or localhost. To filter which equipment is allowed to connect to to WebRTC gateway you can use following new option mpconfig --SSIP=IP_OF_PBX to restrict access to port 5060.
in case of OXE with spatial redundancy use mpconfig --SSIP=IP_1_OF_PBX,IP_2_OF_PBX

1.75.5-302
- Bug fixes:

- Fixed issue VoIP calls not working (e.g. incoming call pop-up is not presented or outgoing calls are not working) (issue in the internal user database of the WebRTC Gateway).
- 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.

1.75.3-296
- Enhancements:

- added VirtualBox VMDK compatibility: do not use mpugrade, use /opt/mediapillar/cv/mp-upgrade.sh --yes --now --nokexec

- Bug fixes:

- Fixed issue no audio when PBX codec is G711u.
- 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).

- Serviceability improvements and enhancements:

- ensure log files are not empty for rotation.
- component upgrade janus to 0.8.2.
- component upgrade libnice to current master 0.1.16.0-3.
- component update nodejs 8 dependencies.

1.74.16-279
- Enhancements:

- Added SIP-TLS signaling between OXE and the WebRTC GW (strict SRTP). Command to activate mpsiptls [on|off].

- Bug fixes:

- Fixed issue users being unregistered after janus crash (missing call notification).
- 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.

- Serviceability improvements and enhancements:

- hints added to mpstatus in case of authentication error.
- 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).

1.74.10-234
- Bug fixes:

- Fixed issue about audio could not be established, due to failing codec negotiation. A mechanism of "renegotiation" is now available.
- 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.

- Serviceability improvements and enhancements:

- log level has been changed to default 0 to avoid log flood leading to WebRTC overload.
- 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.

1.73.17-178
- Serviceability improvements and bug fixes:

- Unidirectional audio and bidirectional no-audio problems have been fixed.
- Some call cuts issues have been fixed.
- mpcheck STUN/TURN tests have been reworked.

1.73.13-155
- 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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