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

Tue Feb 16 19:15:42 MSK 2016

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>
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*
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