freeswitch receive rtp audio

感谢sevendu,

mod_sofia 有个模块rtp.c, 配置文件，加入local_addr等变量即可。

<extension name="Local_Extension">
<condition field="${switch_r_sdp}" expression="(.*)Playback(.*)" break="on-true">
<action application="log" data="CRIT playback"/>
<action application="lua" data="playbackRecord.lua"/>
</condition>
    <condition field="destination_number" expression="^(10[01][0-9])$">
<action application="export" data="dialed_extension=$1"/>
<!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> -->
<action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/>
<action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
<action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/>
<action application="bind_meta_app" data="4 b s execute_extension::att_xfer XML features"/>
<action application="set" data="ringback=${us-ring}"/>
<action application="set" data="transfer_ringback=$${hold_music}"/>
<action application="set" data="call_timeout=30"/>
<action application="export" data="local_addr=192.168.42.7"/>
<action application="export" data="remote_addr=192.168.6.73"/>
<action application="export" data="local_port=6060"/>
<action application="export" data="remote_port=6066"/>
<action application="export" data="codec=PCMU"/>
<action application="export" data="pt=101"/>
<!-- <action application="set" data="sip_exclude_contact=${network_addr}"/> -->
<action application="set" data="hangup_after_bridge=true"/>
<!--<action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> -->
<action application="set" data="continue_on_fail=true"/>
<action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
<action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
<action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/>
<action application="hash" data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/>
<action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
<!--<action application="export" data="nolocal:rtp_secure_media=${user_data(${dialed_extension}@${domain_name} var rtp_secure_media)}"/>-->
<action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
<!--action application="answer"/-->
<!--<action application="playback" data="‪E:\FreeSwitchSip\fs1_4a\x64\Debug\recordings\1000_3000_20141124182233.wav"/>-->
<!--<action application="bridge" data="user/${dialed_extension}@${domain_name}"/>-->
<action application="bridge" data="rtp/123/456"/>
<action application="sleep" data="1000"/>
<action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/>
      </condition>
    </extension>