Asterisk Switching Bridge To Native_rtp Even With Direct_media=no

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Asterisk Users 6 Comments

Hey guys,

have issues with reinvite, no matter what endpoint is calling asterisk always tries switch simple_bridge to native_rtp

Bridge 0422bfa0-9d22-4bba-9108-a3f14d7d1cab: switching from simple_bridge technology to native_rtp

in endpoints table “direct_media” sets to “no” on all endpoints but it doesn’t help.

if native_rtp not work for some reason I have oneway audio. how can I fix this? if I add mix_monitor it works, but it’s not a right way to fix this issues.

Asterisk 13.2.0

6 thoughts on - Asterisk Switching Bridge To Native_rtp Even With Direct_media=no

  • A native_rtp bridge is used for more than direct media. It is also used for local native bridging, that is, when you have two RTP capable channels in a bridge and Asterisk does not require the media to flow through its core. The bridge then just performs a packet to packet swap between the two RTP capable channels.

    Note that on verbosity 4, Asterisk will tell you if the bridge is locally or remotely bridging the two channels.

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    445 Jan Davis Drive NW – Huntsville, AL 35806 – USA
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  • Local (packet to packet) bridging should not do that. Remote (direct media) can do that.

    Can you confirm – by looking at a verbose level 4 log – how Asterisk is bridging the two channels?

  • NAT endpoint calling local endpount – switching to native_rtp then no audio, both of them have direct_media=no, Verbose log:

    — Executing [99@dialmap:1] AGI(“PJSIP/304-00000022”, “/pbx/agi.php”) in new stack
    — Launched AGI Script /pbx/agi.php
    — AGI Script Executing Application: (Dial) Options: (PJSIP/99/sip:99@192.168.1.73:5060,20)
    — Called PJSIP/99/sip:99@192.168.1.73:5060
    — PJSIP/99-00000023 is ringing
    — PJSIP/99-00000023 answered PJSIP/304-00000022
    — Channel PJSIP/304-00000022 joined ‘simple_bridge’ basic-bridge
    — Channel PJSIP/99-00000023 joined ‘simple_bridge’ basic-bridge
    > Bridge da8840bc-9b71-4ca6-b1d8-9565bf8e5e28: switching from simple_bridge technology to native_rtp
    > Locally RTP bridged ‘PJSIP/99-00000023’ and ‘PJSIP/304-00000022’ in stack
    > Locally RTP bridged ‘PJSIP/99-00000023’ and ‘PJSIP/304-00000022’ in stack
    > 0x7f4b50145420 — Probation passed – setting RTP source address to 194.204.157.200:8972
    > 0x7f4b5014f140 — Probation passed – setting RTP source address to 192.168.1.73:5004
    — Channel PJSIP/304-00000022 left ‘native_rtp’ basic-bridge
    — Channel PJSIP/99-00000023 left ‘native_rtp’ basic-bridge
    — AGI Script /pbx/agi.php completed, returning 4

  • Correct – and per the log, they shouldn’t be in a direct media bridge:

    > Locally RTP bridged ‘PJSIP/99-00000023’ and
    ‘PJSIP/304-00000022’ in stack
    > Locally RTP bridged ‘PJSIP/99-00000023’ and
    ‘PJSIP/304-00000022’ in stack

    Locally RTP bridged means media is still flowing through Asterisk, it just isn’t being decoded and passed through the core.

  • Ok, if this is normal why I have oneway audio when nat endpoint calling to local. if mixmonitor or srtp is enabled audio is ok. Issues with native_rtp for sure

    Sent from my iPhone