Freeswitch Module
From OpenSimulator
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| − | The FreeSwitch module enables voice in opensim with no changes required to the SL or Hippo clients (must be | + | The FreeSwitch module enables voice in opensim with no changes required to the SL or Hippo clients (must be between 1.22 and 1.23.5 for Linden Labs Second Life viewer - 2.x viewers do not work at present - and 0.5 or after for Hippo). |
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| + | A little history on the development of the FreeSWITCH module is given here: | ||
* http://zaki.asia/2009/04/28/freeswitch-module-in-opensim/ | * http://zaki.asia/2009/04/28/freeswitch-module-in-opensim/ | ||
Revision as of 12:34, 20 August 2010
The FreeSwitch module enables voice in opensim with no changes required to the SL or Hippo clients (must be between 1.22 and 1.23.5 for Linden Labs Second Life viewer - 2.x viewers do not work at present - and 0.5 or after for Hippo).
A little history on the development of the FreeSWITCH module is given here:
Contents |
FreeSwitch Install
Follow the instructions here on how to compile from source. We need to enable two specific modules. Please ensure you compile from the freeswitch trunk for now until we can post a minimum version number (there are known issues with older versions)
When you get to the part in the instructions where it says "Edit modules.conf so that it will build the modules you desire." edit the modules.conf file and uncomment to make active the entries for xml_curl and the siren14 codec
codecs/mod_siren and xml_int/mod_xml_curl
FreeSwitch Config
Install and compile Freeswitch, making sure you enable the xml_curl module and also the siren14 codec.
enable mod_xml_curl
Next, do not forget to activate mod_xml_curl in /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml. mod_xml_curl is disabled by default on fresh install.
uncomment the lines...
<load module="mod_xml_curl"/>
and
<load module="mod_siren"/>
configure mod_xml_curl
The xml_curl module configuration should point to an opensim region that has the Freeswitch voice module enabled (voice also needs to be enabled in the estate setting for all regions you wish to be supported).
example xml_curl.conf.xml found in /usr/local/freeswitch/conf/autoload_configs
<configuration name="xml_curl.conf" description="cURL XML Gateway">
<bindings>
<binding name="example">
<param name="gateway-url" value="http://youropensimregionip:9000/api/freeswitch-config" bindings="directory"/>
<param name="gateway-credentials" value="freeswitch:password"/>
<param name="disable-100-continue" value="true"/>
</binding>
<binding name="local">
<param name="gateway-url" value="http://youropensimregionip:9000/api/freeswitch-config" bindings="dialplan"/>
<param name="gateway-credentials" value="freeswitch:password"/>
<param name="disable-100-continue" value="true"/>
</binding>
</bindings>
</configuration>
The /usr/local/freeswitch/conf/vars.xml requires modification to enable the siren14 codec
search within vars.xml for the global_codec_prefs and change the line to read
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G722,PCMU,PCMA,GSM"/>
or
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/>
G7221@32000h is the siren14 codec
configure conference.conf.xml
By default, FreeSwitch plays hold music when there is only one avatar in the conference and beeps for everyone when avatars arrive and leave. To disable, edit /usr/local/freeswitch/conf/autoload_configs/conference.conf.xml. Locate the "default" profile and comment out the following lines as shown below:
[...]
<!-- File to play if you are alone in the conference -->
<!-- <param name="alone-sound" value="conference/conf-alone.wav"/> -->
[...]
<!-- File to play when you're alone (music on hold)-->
<!-- <param name="moh-sound" value="$${hold_music}"/> -->
<!-- File to play when you join the conference -->
<!-- <param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/> [^] -->
<!-- File to play when you leave the conference -->
<!-- <param name="exit-sound" value="tone_stream://%(500,0,300,200,100,50,25)"/> [^] -->
<!-- File to play when you ae ejected from the conference -->
<!-- <param name="kicked-sound" value="conference/conf-kicked.wav"/> -->
[...]
OpenSim Config
Add the following section to OpenSim.ini. You will also need to enable voice in the regions estate settings. Make sure the freeswitch server is started BEFORE bringing the region up.
[FreeSwitchVoice] enabled = true ; FreeSwitch server is going to contact us and ask us all sorts of things. freeswitch_server_user = freeswitch freeswitch_server_pass = password freeswitch_api_prefix = /api ; IP address of an opensim region with voice enabled freeswitch_service_server = youropensimregionip ; the port your Opensim region is running on freeswitch_service_port = 9000 ; your freeswitch IP address freeswitch_realm = yourfreeswitchserverip freeswitch_sip_proxy = yourfreeswitchserverip:5060 ; STUN = Simple Traversal of UDP through NATs ; See http://wiki.freeswitch.org/wiki/NAT_Traversal freeswitch_attempt_stun = false freeswitch_stun_server = stun.freeswitch.org ; Echo server MAY not used but information MAY be ; required to relay to Freeswitch server (TBC?) freeswitch_echo_server = yourfreeswitchserverip freeswitch_echo_port = 50505 freeswitch_well_known_ip = yourfreeswitchserverip freeswitch_default_timeout = 5000 freeswitch_subscribe_retry = 120 ; freeswitch_password_reset_url = ; opensim_well_known_http_address = youropensimregionip ; CHECK THIS... should be Address_Of_Your_SIM_HTTP_Server_Hostname_Allowed
Note that if you are running in Opensim grid mode, a single Freeswitch service on yourfreeswitchserverip can support multiple regions on one or more Opensim.exe running on one or more hosts. And the Freeswitch server need not be on the same host as the Opensim.exe used to anchor the connection.
Note STUN server "stun.freeswitch.org" is an example only, and its availability should not be relied upon for a service.
Firewall Config
Make sure that the ports used by Freeswitch are acessible though your firewall(s). In the example above the main port used is 5060 though other ports may be used for call initiation (port 1720 for H.323 Call Signaling) and dynamically assigned ports for specific call traffic. The usual H.323 and SIP traffic dynamic port handling on modern firewalls usually enables this.
Is port 50505 actually used, or is it just something that must be reported by OpenSim when the Freeswitch service makes contact?
More details on the Freeswitch ports used and firewall configuration details are at http://wiki.freeswitch.org/wiki/Firewall
If you have problems with NAT/Routers look at using STUN (Simple Traversal of UDP through NATs). More information at http://wiki.freeswitch.org/wiki/NAT_Traversal
Errors & Solutions
NOTE: FreeSwitch is not a core module and help may not be forthcoming to handle issues.
--[ERR] mod_xml_curl.c:230 xml_url_fetch() Received HTTP error 0 trying to fetch: To solve this error, get the last GIT of FreeSwitch, follow this instructions (for Ubuntu) here.
--mod_event_socket.c -> Socket Error Could not listen on 127.0.0.1:5060: This means something is using the port 5060, usually freeswitch is already running, on Ubuntu, try the following command: netstat -npl | grep 85060 If freeswitch is running, close is with: sudo freeswitch -stop OR sudo invoke-rc.d freeswitch stop
MAKE SURE the OpenSim region you point Freeswitch at IS NOT running when you start FreeSwitch.
ALSO make sure that you have enabled voice in the "About Land"->"Voice" dialogue in the region, you will need to be a region admin to do this.
Check the plot also has voice enabled. You must be a plot owner or have suitable permisison to do this. Recent Linden Labs Second Life viewers do not offer this option, so use Hippo or another client which does.
