[Freeswitch-users] problems with bridging a call, looks like transcoding is disabled

Roman Kudinov roman.kudinov at novelapp.com
Tue Feb 16 20:53:41 MSK 2016


Hi Brian,

I saw this variable but looks like used it the wrong way. Thanks for the 
help, the codecs negotiation works properly now!


16.02.2016 19:16, Brian West:
> This topic was talked about on the list in the past week: 
> https://freeswitch.org/jira/browse/FS-8321
>
> "BEHAVIOR CHANGE Add variable media_mix_inbound_outbound_codecs to mix 
> inbound and outbound codecs"
>
> /b
>
> On Tue, Feb 16, 2016 at 6:40 AM, Roman Kudinov 
> <roman.kudinov at novelapp.com <mailto:roman.kudinov at novelapp.com>> wrote:
>
>     Hi all,
>
>     I have a problem with bridging a call. My FS 1.6.6 is setup to
>     bridge calls from RTMP-based source (using mod_rtmp) to a SIP gateway.
>     I have two branches in the dial plan.
>
>     1) One works through mod_conference which calls an outbound number
>     using conference_set_auto_outcall
>
>     2) Another works by the direct bridging of incoming rtmp call into
>     outbound SIP call.
>
>     Whilst the first branch works just fine, the second one does not.
>     They both use the same sofia profiles, SIP gateways and outbound
>     SIP numbers.
>     They both are called from the same RTMP source. Here are the
>     snippet of codes.
>
>     ================== This one works ====================================
>     > <extension name="conference_set_auto_outcall">
>     >    <condition field="destination_number" expression="123">
>     >      <action application="answer"/>
>     >      <action application="set"
>     data="conference_auto_outcall_caller_id_name=$${effective_caller_id_name}"/>
>     >      <action application="set"
>     data="conference_auto_outcall_caller_id_number=$${effective_caller_id_number}"/>
>     >      <action application="set"
>     data="conference_auto_outcall_profile=default"/>
>     >      <action application="conference_set_auto_outcall"
>     data="{ignore_early_media=true}sofia/gateway/sip_profile/number"/>
>     >      <action application="conference"
>     data="$1+flags{moderator|endconf|mute}"/>
>     >    </condition>
>     > </extension>
>     ===================================================
>
>     ================ This one does not work ================
>     > <extension name="phone_only_session">
>     >    <condition field="destination_number" expression="456">
>     >      <action application="set" data="ignore_early_media=true"/>
>     >      <action application="set"
>     data="absolute_codec_string=PCMU,PCMA,opus"/>
>     >      <action application="bridge"
>     data="sofia/gateway/sip_profile/number"/>
>     >    </condition>
>     > </extension>
>     ========================
>
>     I'd like to outline that they use the same SIP profiles, they are
>     called from the same RTMP-source (they differs by the
>     destination_number), they
>     call the same SIP number.
>     I turned on SIP tracing on and found that the call that is
>     initiated by mod_conference offers the codecs according to
>     outbound_codec_prefs set
>     in vars.xml, here is the piece of log:
>     >      m=audio 18684 RTP/AVP 0 102 103 104 105 8 101 106 108 110
>     >      a=rtpmap:0 PCMU/8000
>     >      a=rtpmap:102 SPEEX/8000
>     >      a=rtpmap:103 SPEEX/16000
>     >      a=rtpmap:104 SPEEX/32000
>     >      a=rtpmap:105 opus/48000/2
>     >      a=fmtp:105 useinbandfec=1; maxaveragebitrate=30000;
>     maxplaybackrate=48000; ptime=20; minptime=10; maxptime=40
>     >      a=rtpmap:8 PCMA/8000
>     >      a=rtpmap:101 telephone-event/8000
>     >      a=fmtp:101 0-16
>     >      a=rtpmap:106 telephone-event/16000
>     >      a=fmtp:106 0-16
>     >      a=rtpmap:108 telephone-event/32000
>     >      a=fmtp:108 0-16
>     >      a=rtpmap:110 telephone-event/48000
>     >      a=fmtp:110 0-16
>     >      a=ptime:20
>
>     But the directly bridged call offers incoming codec only, e.g. speex
>     >      m=audio 24972 RTP/AVP 102 101
>     >      a=rtpmap:102 SPEEX/16000
>     >      a=rtpmap:101 telephone-event/16000
>     >      a=fmtp:101 0-16
>     >      a=ptime:20
>
>     I tried everything I could imagine. I set absolute_codec_string in
>     the dialplan (you can see it in the above snippet).
>     I explicitly set
>     > <X-PRE-PROCESS cmd="set"
>     data="outbound_codec_prefs=PCMU,G722,OPUS,PCMA"/>
>     in vars.xml
>
>     I tried to change
>     > <param name="inbound-codec-negotiation" value="generous"/>
>     from generous to greedy
>
>     I tried with true/false in the following parameters in
>     internal.xml and
>     external.xml SOFIA profiles
>     > <param name="inbound-late-negotiation" value="false"/>
>     > <param name="inbound-zrtp-passthru" value="false"/>
>     Nothing changes.
>
>     Moreover the direct bridging worked fine on FS 1.4.7, it offered
>     PCMU and Speex codecs to SIP.
>     I've upgraded to FS 1.6.6 and now it doesn't work. It looks like I
>     missed an important setting that makes "bridge" application to work in
>     proxy media mode.
>     I checked I don't have neither bypass or proxy words in vars.xml,
>     sofia.conf.xml, internal.xml, external.xml, public.xml or they are
>     commented.
>
>     Does anybody have any ideas about the reason for such behavior?
>
>
>     Thanks,
>     Roman
>
>
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>
> -- 
>
> */Brian West/*
> brian at freeswitch.org <mailto:brian at freeswitch.org>
>
>
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