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About

Sofia is a 

Fs
module (mod_sofia) that provides SIP connectivity to and from FreeSWITCH in the form of a User Agent.

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To understand how to write dialplans, use pattern matching, etc., see Dialplan

The Relationship Between SIP Profiles and Domains

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FreeSWITCH can also subscribe to receive notification of events from the gateway. For more information see Presence#Use_Presence - Use FreeSWITCH _ as _ a _clientClient

Parameters

The following is a list of param elements that are children of a gateway element:

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Where it will first check the specific XML file, then hit normal XML which also do a mod_xml_curl lookup assuming you have that configured and working.

Media related options

See also: Proxy Media

resume-media-on-hold

When calls are in no media this will bring them back to media when you press the hold button. To return the calls to bypass-media after the call is unheld, enable bypass-media-after-hold.

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Codecs related options

Also see:

inbound-codec-prefs

This parameter allows to change the allowed inbound codecs per profile.

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  • 'generous' permits the remote codec list have precedence and 'win' the codec negotiation and selection process
  • 'greedy' forces a win by the local FreeSWITCH preference list
  • 'scrooge' takes 'greedy' a step further, so that the FreeSWITCH wins even when the far side lies about capabilities during the negotiation process

sip_codec_negotiation is a channel variable version of this setting

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When receiving a REGISTER or INVITE, enable NAT mode automatically if IP address in Contact header matches an entry defined in the RFC 1918 access list. "acl" is a misnomer in this case because access will not be denied if the user's contact IP doesn't match.

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VAD and CNG

VAD stands for Voice Activity Detector. FreeSWITCH is capable of detecting speech and can stop transmitting RTP packets when no voice is detected.

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This will force FreeSWITCH to send SIP responses to the network port from which they were received. Use at your own risk! For more information see NAT Traversal.

<param name="NDLB-force-rport" value="true|safe"/>

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This goes in the "..sip_profiles/external.xml" file.

TLS

Please make sure to read http://wiki.freeswitch.org/wiki/SIP_TLS#Limitations_of_the_Freeswitch_TLS.2FSSLv23_Implementation before read SIP TLS before enabling certain features below as they may not behave as expected.

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Note the "_" instead of "-" in profile param (this is var set in dialplan). (24.10.2010: "both" don't seem to me work in my tests, "outbound" does)

Note: for inband DTMF, Misc. Dialplan Tools start_dtmf must be used in the dialplan.
Also, to change the outgoing routing from info or rfc2833 to inband, use Misc._Dialplan_Tools_start_dtmf_generate

RFC 2833

pass-rfc2833

Default: false

If true, it passes RFC 2833 DTMF's from one side of a bridge to the other, untouched. Otherwise, it decodes and re-encodes them before passing them on.

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Set this to the size of the jitterbuffer you would like to have on all calls coming through this profile.

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Can be set to 'true', 'false' or 'first-only'. If set to 'true' (this is the default behavior), mod_sofia will send a message-query event upon registration. mod_voicemail uses this for counting messages.

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unregister-on-options-fail

If set to True with nat-options-ping the endpoint will be unregistered if no answer on OPTIONS packet.

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With this option set FreeSWITCH will periodically send an OPTIONS packet to all NATed registered endpoints to keep alive connection. If set to True with unregister-on-options-fail the endpoint will be unregistered if no answer on OPTIONS packet.

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With this option set FreeSWITCH will periodically send an OPTIONS packet to all registered endpoints to keep alive connection. If set to True with unregister-on-options-fail the endpoint will be unregistered if no answer on OPTIONS packet.

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Default Configuration File

From Git

Reloading

If you've made a change in sofia.conf.xml, there are two ways to get FreeSWITCH to use the new values.

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  • sofia profile <profilename> restart reloadxml

Security Features

  • SIP TLS for secure signaling.
  • SRTP for secure media delivery.
  • The Auth section above for authentication settings.