|Patrik Dahlström ee76a2e854 Setup a SIGINT handler||8 months ago|
|media||3 years ago|
|.gitignore||5 years ago|
|AudioFramesMixer.cpp||5 years ago|
|AudioFramesMixer.hpp||5 years ago|
|CMakeLists.txt||8 months ago|
|Configuration.cpp||3 years ago|
|Configuration.hpp||3 years ago|
|IncomingConnectionValidator.cpp||8 months ago|
|IncomingConnectionValidator.hpp||5 years ago|
|LICENSE||5 years ago|
|MumbleChannelJoiner.cpp||3 years ago|
|MumbleChannelJoiner.hpp||3 years ago|
|MumbleCommunicator.cpp||3 years ago|
|MumbleCommunicator.hpp||3 years ago|
|PjsuaCommunicator.cpp||3 years ago|
|PjsuaCommunicator.hpp||3 years ago|
|README.md||3 years ago|
|config.ini.example||3 years ago|
|main.cpp||8 months ago|
|main.hpp||3 years ago|
|make-client-certs.sh||3 years ago|
SIP to Mumble gateway based on PJSIP stack and mumlib library. It registers to SIP registrar and listens for incoming connections on the SIP account.
This enables the user to participate in Mumble conference using SIP client or perhaps ordinary telephone, by VoIP provider.
Install all needed dependencies
Clone and compile Mumlib library. Since it doesn't have any installer, clone it to common directory:
mkdir mumsi-dist && cd mumsi-dist git clone https://github.com/slomkowski/mumlib.git mkdir mumlib/build && cd mumlib/build cmake .. make cd -
Then clone and build mumsi:
git clone https://github.com/slomkowski/mumsi.git mkdir mumsi/build && cd mumsi/build cmake .. make
Copy example config.ini file and edit it according to your needs:
cp config.ini.example config.ini
Remember to add URIs which you want to make calls from. Calls from other URIs won't be answered.
To run the service, type:
If your SIP provider allows multiple simultaneous calls, mumsi can be configured to accept calls and map them to separate Mumble users. The max_calls is configure in config.ini:
[sip] ... max_calls = 32 ...
Currently, the Mumble connections are established at server start. For usability, the following options are recommended:
The maximum number of calls is set in main.hpp and should not exceed the PJSUA_MAX_CALLS in pjsua.h, which by default is 32. This can also be recompiled to more, if desired.
When mumsi logs into Mumble, it uses the user name from config.ini and appends the character '-', followed by the connection number (counter).
LIMITATIONS: The code is alpha and needs testing/debugging, especialy in the code that uses mumlib::Transport. Also, there is initial work on connecting the Mumble user only when the SIP call is active, so the UI for other users is better, but this code is still very buggy and therefore disabled.
When the caller_pin is set, the incoming SIP connection is mute/deaf until the caller enters the correct PIN, followed by the '#' symbol. On three failed attempts, the SIP connection is hung up. On success, the Mumble user is moved into the channel matching channelAuthExpression, if specified, and then mute/deaf is turned off. As a courtesy to the other users, a brief announcement audio file is played in the Mumble channel.
The caller_pin is configured in config.ini in the app section:
[app] caller_pin = 12345
In addition to the caller_pin, a channelAuthExpression can be set. After the caller authenticates with the PIN, the mumsi Mumble user will switch to the Mumble channel that matches this expression. When the call is completed, the mumsi Mumble user will return to the default channel that matches channelNameExpression.
This helps keep the unused SIP connections from cluttering your channel.
By default (i.e. autodeaf=0), other Mumble users can only see whether the mumsi connection has an active caller if they are in the same channel. This is becaue the 'talking mouth' icon is not visible to users in other channels. The mute/deaf icons, on the other hand, can be seen by Mumble users when they are in different channels, making it easier to spot when a new caller has connected.
Setting `autodeaf=1' causes the mumsi Mumble user to be mute/deaf when there is no active SIP call.
When certain events occur, it is user-friendly to provide some sort of prompting confirmation to the user. An example set of WAV files is provided, but they can easily be customized or replaced with local versions, if needed. If the files are not found, no sound is played. The following events are supported:
mumsi provides no init.d scripts, but you can use great daemon mangaer, Supervisor. The sample configuration file:
[program:mumsi] command=/home/mumsi/mumsi-dist/mumsi/build/mumsi config.ini directory=/home/mumsi/mumsi-dist/mumsi user=mumsi stdout_logfile=/home/mumsi/console.log stdout_logfile_maxbytes=1MB stdout_logfile_backups=4 stdout_capture_maxbytes=1MB redirect_stderr=true
Remember to allow incoming connections on port 5060 UDP in your firewall. If you're connecting to public SIP provider from machine behind NAT, make sure your setup works using some generic SIP client. Since SIP is not NAT-friendly by design, PJSIP usually takes care of connection negotiation and NAT traversal, but might fail. The most reliable solution is to configure port forwarding on your home router to your PC.
You may encounter following error when running mumsi on older distros
pjsua_conf_add_port(mediaPool, (pjmedia_port *)port, &id) error: Invalid operation (PJ_EINVALIDOP)
Some older versions of PJSIP are affected (confirmed for 2.3). In this case you have to update PJSIP to most recent version (2.4.5).
The multi-caller code is alpha and needs testing/debugging, especialy in the code that uses mumlib::Transport. Also, there is initial work on connecting the Mumble user only when the SIP call is active, so the UI for other users is better, but this code is still very buggy and therefore disabled.
2015, 2016 Michał Słomkowski. The code is published under the terms of Apache License 2.0.
If this project has helped you and you feel generous, you can donate some money to