Sofia
From FreeSWITCH Wiki
Introduction
Sofia is a SIP stack used by FreeSWITCH. When you see "sofia" anywhere in your configuration, think "This is SIP stuff." It takes a while to master it all, so please be patient with yourself. SIP is a crazy protocol and it will make you crazy too if you aren't careful. Read on for information on setting up SIP/Sofia in your FreeSWITCH configuration.
Sofia Configuration Files, sofia.conf.xml
See Sofia Configuration Files.
What if these commands don't work for me?
Make sure that you are not running another SIP server at the same time as FreeSWITCH. It is not always obvious that another SIP server is running. If you type in Sofia commands such as 'sofia status profile default' and it doesn't work then you may have another SIP server running. Stop the other SIP server and restart FreeSWITCH.
Flushing/rebooting registered endpoints
You can flush a registration or reboot specific registered endpoint by issuing a flush_inbound_reg command from the console
freeswitch> sofia profile <profile_name> flush_inbound_reg [<call_id>|<user@host>] [reboot]
If you leave out <call_id> and/or <user@host>, you will flush/reboot every registered endpoint on a profile.
Reloading profiles and gateways
You can reload a specific SIP profile by issuing a rescan/restart command from the console
freeswitch> sofia profile <profile_name> [<rescan>|<restart>] reloadxml
The difference between rescan and restart is that rescan will just load new config and not stop FS from processing any more calls on a profile. Some config options like IP and port are not reloaded with rescan.
Deleting gateways
You can delete a specific gateway by issuing a killgw command from the console
freeswitch> sofia profile <profile_name> killgw <gateway_name>
Restarting gateways
You can force a gateway to restart ( good for forcing a re-registration or similar ) by issuing a killgw command from the console followed by a profile rescan. this is safe to perform on a profile that has active calls.
freeswitch> sofia profile <profile_name> killgw <gateway_name> freeswitch> sofia profile <profile_name> rescan
Adding / Changing Existing Gateways
It will be assumed that you have all your gateways in the /usr/local/freeswitch/conf/sip_profiles/external directory and that you have just created a new entry. You can add a new gateway to FreeSwitch by issuing a rescan reloadxml command from the console as seen in the example below. This will load the newly created gateway and not affect any calls that are currently up.
freeswitch> sofia profile external rescan reloadxml
You now realize that you have screwed up the IP address in the new gateway and need to change it. So you edit your gateway file and make any changes that you want. You will then need to issue the following commands to destroy the gateway, and then have FreeSwitch reload the changes with affecting any existing calls that are currently up.
freeswitch> sofia profile external killgw <gateway_name> freeswitch> sofia profile external rescan reloadxml
View SIP Registrations
You can view all the devices that have registered by running the following from the console
freeswitch> sofia status profile <profile name> freeswitch> sofia status profile default freeswitch> sofia status profile outbound
You can also use the xmlstatus key to retrieve statuses in XML format. This is specially useful if you are using mod_xml_rpc.
Commands are as follows:
freeswitch> sofia xmlstatus profile <profile name> freeswitch> sofia xmlstatus profile default freeswitch> sofia xmlstatus profile outbound
Debugging Sofia-SIP
The Sofia-SIP components can output various debugging information. The detail of the debugging output is determined by the debugging level. The level is usually module-specific and it can be modified by module-specific environment variable. There is also a default level for all modules, controlled by environment variable #SOFIA_DEBUG.
The environment variables controlling the logging and other debug output are as follows:
- #SOFIA_DEBUG Default debug level (0..9) - #NUA_DEBUG User Agent engine (<a href="nua/index.html">nua</a>) debug level (0..9) - #SOA_DEBUG SDP Offer/Answer engine (<a href="soa/index.html">soa</a>) debug level (0..9) - #NEA_DEBUG Event engine (<a href="nea/index.html">nea</a>) debug level (0..9) - #IPTSEC_DEBUG HTTP/SIP autentication module debug level (0..9) - #NTA_DEBUG Transaction engine debug level (0..9) - #TPORT_DEBUG Transport event debug level (0..9) - #TPORT_LOG If set, print out all parsed SIP messages on transport layer - #TPORT_DUMP Filename for dumping unparsed messages from transport - #SU_DEBUG <a href="nea/index.html">su</a> module debug level (0..9)
The defined debug output levels are:
- 0 SU_DEBUG_0() - fatal errors, panic - 1 SU_DEBUG_1() - critical errors, minimal progress at subsystem level - 2 SU_DEBUG_2() - non-critical errors - 3 SU_DEBUG_3() - warnings, progress messages - 5 SU_DEBUG_5() - signaling protocol actions (incoming packets, ...) - 7 SU_DEBUG_7() - media protocol actions (incoming packets, ...) - 9 SU_DEBUG_9() - entering/exiting functions, very verbatim progress
Starting with 1.0.4, those parameters can be controlled from the console by doing
freeswitch> sofia loglevel <all|default|tport|iptsec|nea|nta|nth_client|nth_server|nua|soa|sresolv|stun> [0-9]
"all" Will change every component's loglevel
A log level of 0 turns off debugging, to turn them all off, you can do
freeswitch> sofia loglevel all 0
Sample Export (Linux/UNIX)
Alternatively, the levels can also be read from environment variables. The following bash commands turn on all debugging levels, and is equivalent to "sofia loglevel all 9"
export SOFIA_DEBUG=9 export NUA_DEBUG=9 export SOA_DEBUG=9 export NEA_DEBUG=9 export IPTSEC_DEBUG=9 export NTA_DEBUG=9 export TPORT_DEBUG=9 export TPORT_LOG=9 export TPORT_DUMP=/tmp/tport_sip.log export SU_DEBUG=9
To turn this debugging off again, you have to exit FreeSWITCH and type unset. For example:
unset TPORT_LOG
Sample Set (Windows)
The following bash commands turn on all debugging levels.
set SOFIA_DEBUG=9 set NUA_DEBUG=9 set SOA_DEBUG=9 set NEA_DEBUG=9 set IPTSEC_DEBUG=9 set NTA_DEBUG=9 set TPORT_DEBUG=9 set TPORT_LOG=9 set TPORT_DUMP=/tmp/tport_sip.log set SU_DEBUG=9
To turn this debugging off again, you have to exit FreeSWITCH and type unset. For example:
set TPORT_LOG=
You can also control SIP Debug output within fs_cli, the freeswitch client app.
freeswitch> sofia profile <profilename> siptrace on|off
Profile Configurations
ACL
You can restrict access by IP address for either REGISTERs or INVITEs (or both) by using the following options in the sofia profile.
<param name="apply-inbound-acl" value="<acl_list|cidr>"/> <param name="apply-register-acl" value="<acl_list|cidr>"/>
See Acl for other access controls
See acl.conf.xml for list configuration
Using A Single Domain For All Registrations
You can force all registrations in a particular profile to use a single domain. In other words, you can ignore the domain in the SIP message. You will need to modify several sofia profile settings.
challenge realm
<param name="challenge-realm" value="auto_from"/>
Defaults to "auto_to" if not set; default config specifies "auto_from"
Possible Values:
- auto_from - uses the from field as the value for the SIP realm.
- auto_to - uses the to field as the value for the SIP realm.
- <anyvalue> - you can input any value to use for the SIP realm.
force-register-domain
<param name="force-register-domain" value="$${domain}"/>
This will force the profile to ignore the domain in the to or from packet and force it to the value listed here for this parameter.
This will store the info into the database with the user@<from domain in SIP packet>
force-register-db-domain
<param name="force-register-db-domain" value="$${domain}"/>
This will work in conjunction with force-register-domain so that the forced domain is stored in the database also.
Forcing SIP profile to use a static IP address
The default FreeSWITCH configuration will automatically determine the IP address of your local network interface. If you want to manually set the IP (for example, if you have multiple network interfaces on a single machine), you can set the following parameters in the Sofia profile:
<param name="rtp-ip" value="192.168.1.3"/> <param name="sip-ip" value="192.168.1.3"/> where 192.168.1.3 is the IP you want it to bind to.
and also in the sofia.conf.xml file:
<param name="auto-restart" value="false"/> This will prevent the profile from automatically restarting when it detects a network address change.
Syntax
Call an extension on a Remote SIP Server
Basic form:
sofia/<profile>/<extension>@<remoteserver>
Example 1:
sofia/$${profile}/$1/@example.org
Example 2:
sofia/foo/0@sipphone.com
Where <profile> is the name of one of the profiles defined in sofia.conf.xml. By default, there is one profile defined as name="$${domain}", where the $${domain} variable is defined in freeswitch.xml, and defaults to mydomain.com.
Therefore, if you have not changed these config files and are dialing an extension on a remote server, the config would be
sofia/mydomain.com/<extension>@<remoteserver>
To list all profiles defined, type sofia status on the CLI.
Call a locally registered endpoint
Basic form:
sofia/<profile>/<extension>%<localserver>
Example 1:
sofia/$${profile}/$1%$${domain}
Example 2:
sofia/foo/101%192.168.1.1
where foo is the SIP profile, 101 is the userid of the registered endpoint, and 192.168.1.1 is the IP address of freeswitch.
If your SIP profile name is set to your domain, or the domain attribute is set in the profile (e.g., <profile name="internal" domain="$${domain}">), you can originate with the following:
sofia/<profile>/<extension>
And forego the domain when dialing local extensions.
Multiple Registrations
Call one extension and ring several phones
You must enable multiple registrations in sofia.conf
<param name="multiple-registrations" value="true"/>
The most likely place to put this is in a specific profile.
In your dialplan, call the extension using sofia_contact similar to the following:
<action application="bridge" data="${sofia_contact($${sip_profile}/${dialed_ext}@$${domain})}"/>
sofia_contact also works at the CLI, so you can test before you add it to your dialplan
See Function sofia contact for more informations.
Note (this applies to FS 1.0.1 and later): you can disable multiple registrations on a per-user basis by setting the variable "sip-allow-multiple-registrations" to "false" in the directory. In this case, that single user won't be allowed to use multiple registrations.
Flushing Inbound Registrations
From time to time, you may need to kill a registration.
you can kill a registration from the cli, or anywhere that accepts api commands with a command similar to the following
sofia profile <profile_name_here> flush_inbound_reg [optional_callid]
Dial out of a gateway
Basic form:
sofia/gateway/<gateway>/<number_to_dial>
Example 1:
sofia/gateway/asterlink/18005551212
gateway: is a keyword and not a "gateway" name. It has special meaning and tells the stack which credentials to use when challenged for the call.
<gateway> is the actual name of the gateway through which you want to send the call
Your available gateways (usually configured in conf/sip_profiles/external/*.xml) will show up in sofia status:
freeswitch#> sofia status
Name Type Data State
=================================================================================================
default profile sip:mod_sofia@2.3.4.5:5060 RUNNING (2)
mygateway gateway sip:username@1.2.3.4 NOREG
phonebooth.example.com alias default ALIASED
=================================================================================================
1 profile 1 alias
Modifying the To: header
You can override the To: header by appending ^<toheader>.
Example 1:
sofia/foo/user%192.168.1.1^101@$${domain}
Specifying SIP Proxy With fs_path
You can route a call through a specific SIP proxy by using the "fs_path" directive. Example:
sofia/foo/user@that.domain;fs_path=sip:proxy.this.domain
Channel Variables
You can add arbitary headers to outbound SIP calls by prefixing the string 'sip_h_' to any channel variable, for example:
<action application="set" data="sip_h_X-Answer=42"/> <action application="bridge" data="sofia/mydomain.com/1000@example.com"/>
While not required, you should prefix your headers with "X-" to avoid issues with interoperability with other SIP stacks.
All inbound SIP calls will install any X- headers into local variables. This means you can easily bridge any X- header from one freeswitch instance to another.
Additional Variables
Additional variables may also be set to influence the way calls are handled by sofia.
For example, contacts can be filtered by setting the 'sip_exclude_contact' variable. Example:
<anti-action application="set" data="sip_exclude_contact=${network_addr}"/>
Or you can perform SIP Digest authorization on outgoing calls by setting sip_auth_username and sip_auth_password variables to avoid using Gateways to authentificate. Example:
<action application="bridge" data="[sip_auth_username=secretusername,sip_auth_password=secretpassword]sofia/external/00123456789@sip.example.com"/>
Changing the SIP Contact user FreeSwitch normally uses mod_sofia@ip:port for the internal SIP contact. To change this to foo@ip:port, there is a variable, sip_contact_user:
{sip_contact_user=foo}sofia/my_profile/1234@192.168.0.1;transport=tcp
Transcoding Issues
G729 and G723 will not let you transcode because of licensing issues. Calls will fail if for example originating endpoint has set G729 with higher priority and receiving endpoint has G723 with highest priority. The logic is to fail the call rather than attempt to find a codec match. If you are having issues due to transcoding you may disable transcoding and both endpoints will negotiate the compatible codec rather than just fail the call.
Add the following command <param name="disable-transcoding" value="true"/> to your freeswitch.xml.
Example:
<configuration name="sofia.conf" description="sofia Endpoint">
<profiles>
<profile name="sip">
<settings>
<param name="disable-transcoding" value="true"/>
Custom Events
The following are events that can be subscribed to via Event Socket
- sofia::register
- sofia::expire
- sofia::gateway_state
FAQ
Does it use UDP or TCP?
By default it uses both, but you can add ;tport=tcp to the sofia url to force it to use tcp.
For example:
sofia/profile/foo@bar.com;transport=tcp
Also there is a parameter in the gateway config:
<param name="register-transport" value="tcp"/>
That will cause it to use the tcp transport for the registration and all subsequent SIP messages.
Not sure if this is needed or what it does, but the following can also be used in gateway settings:
<!--extra sip params to send in the contact--> <param name="contact-params" value="tport=tcp"/>
See Also
- Debugging_Freeswitch
- SIP Protocol Messages
- Authentication params for gateways.
- RTP Issues
External links
http://opensource.nokia.com/projects/sofia-sip/
