forked from Mirrors/freeswitch
0ed9ebe492
mod_sofia will now examine a variable in the channel to see what the channel's originator was using for a codec and try to put that to the top of the list in the sdp. if this new sofia profile param is set: <param name="disable-transcoding" value="true"/> All outbound calls will use *only* the codec that thier originator is using to ensure no transcoding. (of course that could lead to a failed call where there is no way to do this, so use sparingly) git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@4073 d0543943-73ff-0310-b7d9-9358b9ac24b2
60 lines
2.4 KiB
XML
60 lines
2.4 KiB
XML
<configuration name="sofia.conf" description="sofia Endpoint">
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<profiles>
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<profile name="$${domain}">
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<registrations>
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<!-- <registration name="asterlink">
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<param name="register-scheme" value="Digest"/>
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<param name="register-realm" value=""/>
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<param name="register-username" value="1001"/>
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<param name="register-password" value="nhy65tgb"/>
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<param name="register-from" value="sip:1001@208.64.200.40"/>
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<param name="register-to" value="sip:1001@conference.freeswitch.org"/>
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<param name="register-proxy" value="sip:conference.freeswitch.org:5060"/>
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<param name="register-frequency" value="20"/>
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</registration> -->
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</registrations>
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<settings>
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<param name="debug" value="1"/>
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<param name="rfc2833-pt" value="101"/>
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<param name="sip-port" value="5060"/>
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<param name="dialplan" value="enum,XML"/>
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<param name="dtmf-duration" value="100"/>
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<param name="codec-prefs" value="$${default_codecs}"/>
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<param name="codec-ms" value="20"/>
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<param name="use-rtp-timer" value="true"/>
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<param name="rtp-timer-name" value="soft"/>
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<param name="rtp-ip" value="auto"/>
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<param name="sip-ip" value="auto"/>
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<!--Uncomment to set all inbound calls to no media mode-->
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<!--<param name="inbound-no-media" value="true"/>-->
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<!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok-->
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<!--<param name="inbound-late-negotiation" value="true"/>-->
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<!-- this lets anything register -->
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<!-- comment the next line and uncomment one or both of the other 2 lines for call authentication -->
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<param name="accept-blind-reg" value="true"/>
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<!--TTL for nonce in sip auth-->
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<param name="nonce-ttl" value="60"/>
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<!--Uncomment if you want to force the outbound leg of a bridge to only offer the codec
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that the originator is using-->
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<!--<param name="disable-transcoding" value="true"/>-->
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<!--<param name="auth-calls" value="true"/>-->
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<!-- on authed calls, authenticate *all* the packets not just invite -->
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<!--<param name="auth-all-packets" value="true"/>-->
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<!-- optional ; -->
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<!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>-->
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<!-- <param name="ext-rtp-ip" value="100.101.102.103"/> -->
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<!-- VAD choose one (out is a good choice); -->
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<!-- <param name="vad" value="in"/> -->
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<!-- <param name="vad" value="out"/> -->
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<!-- <param name="vad" value="both"/> -->
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<!--<param name="alias" value="sip:10.0.1.251:5555"/>-->
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</settings>
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</profile>
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</profiles>
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</configuration>
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