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Hi Brian,<br>
<br>
I saw this variable but looks like used it the wrong way. Thanks for
the help, the codecs negotiation works properly now!<br>
<br>
<br>
<div class="moz-cite-prefix">16.02.2016 19:16, Brian West:<br>
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<blockquote type="cite">
<div class="moz-text-html" lang="x-unicode">
<div dir="ltr">This topic was talked about on the list in the
past week: <a
href="https://freeswitch.org/jira/browse/FS-8321"><a class="moz-txt-link-freetext" href="https://freeswitch.org/jira/browse/FS-8321">https://freeswitch.org/jira/browse/FS-8321</a></a>
<div><br>
</div>
<div>"BEHAVIOR CHANGE Add variable
media_mix_inbound_outbound_codecs to mix inbound and
outbound codecs"</div>
<div><br>
</div>
<div>/b</div>
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<div class="gmail_extra"><br>
<div class="gmail_quote">On Tue, Feb 16, 2016 at 6:40 AM,
Roman Kudinov <span dir="ltr"><<a
href="mailto:roman.kudinov@novelapp.com" target="_blank"><a class="moz-txt-link-abbreviated" href="mailto:roman.kudinov@novelapp.com">roman.kudinov@novelapp.com</a></a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">Hi all,<br>
<br>
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.<br>
I have two branches in the dial plan.<br>
<br>
1) One works through mod_conference which calls an
outbound number using conference_set_auto_outcall<br>
<br>
2) Another works by the direct bridging of incoming rtmp
call into outbound SIP call.<br>
<br>
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.<br>
They both are called from the same RTMP source. Here are
the snippet of codes.<br>
<br>
================== This one works
====================================<br>
> <extension name="conference_set_auto_outcall"><br>
> <condition field="destination_number"
expression="123"><br>
> <action application="answer"/><br>
> <action application="set"
data="conference_auto_outcall_caller_id_name=$${effective_caller_id_name}"/><br>
> <action application="set"
data="conference_auto_outcall_caller_id_number=$${effective_caller_id_number}"/><br>
> <action application="set"
data="conference_auto_outcall_profile=default"/><br>
> <action
application="conference_set_auto_outcall"
data="{ignore_early_media=true}sofia/gateway/sip_profile/number"/><br>
> <action application="conference"
data="$1+flags{moderator|endconf|mute}"/><br>
> </condition><br>
> </extension><br>
===================================================<br>
<br>
================ This one does not work ================<br>
> <extension name="phone_only_session"><br>
> <condition field="destination_number"
expression="456"><br>
> <action application="set"
data="ignore_early_media=true"/><br>
> <action application="set"
data="absolute_codec_string=PCMU,PCMA,opus"/><br>
> <action application="bridge"
data="sofia/gateway/sip_profile/number"/><br>
> </condition><br>
> </extension><br>
========================<br>
<br>
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<br>
call the same SIP number.<br>
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<br>
in vars.xml, here is the piece of log:<br>
> m=audio 18684 RTP/AVP 0 102 103 104 105 8 101
106 108 110<br>
> a=rtpmap:0 PCMU/8000<br>
> a=rtpmap:102 SPEEX/8000<br>
> a=rtpmap:103 SPEEX/16000<br>
> a=rtpmap:104 SPEEX/32000<br>
> a=rtpmap:105 opus/48000/2<br>
> a=fmtp:105 useinbandfec=1;
maxaveragebitrate=30000; maxplaybackrate=48000; ptime=20;
minptime=10; maxptime=40<br>
> a=rtpmap:8 PCMA/8000<br>
> a=rtpmap:101 telephone-event/8000<br>
> a=fmtp:101 0-16<br>
> a=rtpmap:106 telephone-event/16000<br>
> a=fmtp:106 0-16<br>
> a=rtpmap:108 telephone-event/32000<br>
> a=fmtp:108 0-16<br>
> a=rtpmap:110 telephone-event/48000<br>
> a=fmtp:110 0-16<br>
> a=ptime:20<br>
<br>
But the directly bridged call offers incoming codec only,
e.g. speex<br>
> m=audio 24972 RTP/AVP 102 101<br>
> a=rtpmap:102 SPEEX/16000<br>
> a=rtpmap:101 telephone-event/16000<br>
> a=fmtp:101 0-16<br>
> a=ptime:20<br>
<br>
I tried everything I could imagine. I set
absolute_codec_string in the dialplan (you can see it in
the above snippet).<br>
I explicitly set<br>
> <X-PRE-PROCESS cmd="set"
data="outbound_codec_prefs=PCMU,G722,OPUS,PCMA"/><br>
in vars.xml<br>
<br>
I tried to change<br>
> <param name="inbound-codec-negotiation"
value="generous"/><br>
from generous to greedy<br>
<br>
I tried with true/false in the following parameters in
internal.xml and<br>
external.xml SOFIA profiles<br>
> <param name="inbound-late-negotiation"
value="false"/><br>
> <param name="inbound-zrtp-passthru"
value="false"/><br>
Nothing changes.<br>
<br>
Moreover the direct bridging worked fine on FS 1.4.7, it
offered PCMU and Speex codecs to SIP.<br>
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<br>
proxy media mode.<br>
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.<br>
<br>
Does anybody have any ideas about the reason for such
behavior?<br>
<br>
<br>
Thanks,<br>
Roman<br>
<br>
<br>
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<p><font face="courier new, monospace"><b><i><font
size="4">Brian West</font></i></b><br>
<span style="font-size:x-small"><a
href="mailto:brian@freeswitch.org"
target="_blank"><a class="moz-txt-link-abbreviated" href="mailto:brian@freeswitch.org">brian@freeswitch.org</a></a></span></font></p>
<p><font size="1" face="courier new,
monospace"><img shrinktofit="true"
src="cid:part13.04080505.07000900@novelapp.com"><br>
</font></p>
<p><font size="2" face="monospace,
monospace"><b><i>Twitter: @FreeSWITCH
, @briankwest</i></b><br>
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href="http://www.freeswitchbook.com"
target="_blank"><a class="moz-txt-link-freetext" href="http://www.freeswitchbook.com">http://www.freeswitchbook.com</a></a><br>
<a
href="http://www.freeswitchcookbook.com"
target="_blank"><a class="moz-txt-link-freetext" href="http://www.freeswitchcookbook.com">http://www.freeswitchcookbook.com</a></a></font></p>
<p><font face="monospace, monospace">Got
Bugs? Report them <a
href="https://freeswitch.org/jira"
target="_blank">here</a>! | Reddit: <a
href="https://www.reddit.com/r/freeswitch" target="_blank">/r/freeswitch</a></font></p>
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