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

[Roman Kudinov]

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