bigbluebutton-Github/bbb-voice-conference/config/freeswitch/conf/dialplan/public/bbb_sfu.xml

<include>
    <extension name="bbb_webrtc_call" continue="true">
      <condition field="${sip_user_agent}" expression="bbb-webrtc-sfu" break="on-false">
        <action application="set" data="presence_data=from_bbb-webrtc-sfu"/>
        <action application="set" data="bbb_authorized=true"/>
        <action application="set" data="rtp_manual_rtp_bugs=ACCEPT_ANY_PACKETS"/>
        <action application="set" data="jb_use_timestamps=true"/>
        <action application="transfer" data="${destination_number} XML default"/>
      </condition>
    </extension>
</include>
feat: add new dialplan rule for bbb-webrtc-sfu calls This new dialplan rule filters calls originating from bbb-webrtc-sfu via SIP user agent parsing. The default bbb-webrtc-sfu UA is "bbb-webrtc-sfu". A new dialplan rule is needed to force RTP auto-adjustment for calls originating in bbb-webrtc-sfu (rtp_manual_bugs=ACCEPT_ANY_PACKETS). That is due to the fact that bidirectional mediasoup bridging is done via an RTP/AVPF endpoint which does not use ICE. FreeSWITCH arbitrarily blocks off auto adjustment for AVPF profiles (presuming ICE), so it needs to be forced otherwise the bridge won't work properly in all environments. Bridging mediasoup and FS via WebRTC (which would circumvent that) is currently not an option due to the fact that FreeSWITCH doesn't handle STUN role conflicts properly (and there will always be a conflict since the initiator is controlled and FS always defaults to controlled) Briding mediasoup and FS via plain RTP/AVP (which would also circumvent that) is not an option right now due to the fact that FreeSWITCH doesn't make ssrcs public in signaling for RTP/AVP profiles. mediasoup needs the remote ssrcs. This could work by pre-generating a ssrc in bbb-webrtc-sfu, signaling it via a SIP header and then specifying it in the rtp_use_ssrc channel variable in FS, which would allow us to shim the ssrc in FS's answer in bbb-webrtc-sfu. Maybe in the future. 2022-01-28 01:57:13 +08:00			`<include>`
			`<extension name="bbb_webrtc_call" continue="true">`
			`<condition field="${sip_user_agent}" expression="bbb-webrtc-sfu" break="on-false">`
feat: add filterable identifier to FS channels originated from SFU Use the presence_data field to annotate channels with a filterable identifier that allows us to differentiate SIP.js channels from SFU ones. The motivation: allow metrics exporters/instrumentations/etc to generate comparison metrics (eg.: mediaStats, usage) between the default bridge and the experimental one without having to do multiple or overly verbose json_api or mod_command/fs_cli calls to filter channels out. 2022-05-12 21:08:11 +08:00			`<action application="set" data="presence_data=from_bbb-webrtc-sfu"/>`
feat: add new dialplan rule for bbb-webrtc-sfu calls This new dialplan rule filters calls originating from bbb-webrtc-sfu via SIP user agent parsing. The default bbb-webrtc-sfu UA is "bbb-webrtc-sfu". A new dialplan rule is needed to force RTP auto-adjustment for calls originating in bbb-webrtc-sfu (rtp_manual_bugs=ACCEPT_ANY_PACKETS). That is due to the fact that bidirectional mediasoup bridging is done via an RTP/AVPF endpoint which does not use ICE. FreeSWITCH arbitrarily blocks off auto adjustment for AVPF profiles (presuming ICE), so it needs to be forced otherwise the bridge won't work properly in all environments. Bridging mediasoup and FS via WebRTC (which would circumvent that) is currently not an option due to the fact that FreeSWITCH doesn't handle STUN role conflicts properly (and there will always be a conflict since the initiator is controlled and FS always defaults to controlled) Briding mediasoup and FS via plain RTP/AVP (which would also circumvent that) is not an option right now due to the fact that FreeSWITCH doesn't make ssrcs public in signaling for RTP/AVP profiles. mediasoup needs the remote ssrcs. This could work by pre-generating a ssrc in bbb-webrtc-sfu, signaling it via a SIP header and then specifying it in the rtp_use_ssrc channel variable in FS, which would allow us to shim the ssrc in FS's answer in bbb-webrtc-sfu. Maybe in the future. 2022-01-28 01:57:13 +08:00			`<action application="set" data="bbb_authorized=true"/>`
			`<action application="set" data="rtp_manual_rtp_bugs=ACCEPT_ANY_PACKETS"/>`
			`<action application="set" data="jb_use_timestamps=true"/>`
			`<action application="transfer" data="${destination_number} XML default"/>`
			`</condition>`
			`</extension>`
			`</include>`