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


 
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roman.kudinov at novel...
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PostPosted: Tue Feb 16, 2016 10:30 am    Post subject: [Freeswitch-users] problems with bridging a call, looks like Reply with quote

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 ====================================
Quote:
<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 ================
Quote:
<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:
Quote:
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
Quote:
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
Quote:
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,G722,OPUS,PCMA"/>
in vars.xml

I tried to change
Quote:
<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
Quote:
<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


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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brian at freeswitch.org
Guest





PostPosted: Tue Feb 16, 2016 11:17 am    Post subject: [Freeswitch-users] problems with bridging a call, looks like Reply with quote

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@novelapp.com (roman.kudinov@novelapp.com)> wrote:
Quote:
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 ====================================
Quote:
<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 ================
Quote:
<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:
Quote:
      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
Quote:
      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
Quote:
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,G722,OPUS,PCMA"/>
in vars.xml

I tried to change
Quote:
<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
Quote:
<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


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org (consulting@freeswitch.org)
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




--

Brian West
brian@freeswitch.org (brian@freeswitch.org)


Twitter: @FreeSWITCH , @briankwest
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com
Got Bugs? Report them here! | Reddit: /r/freeswitch
T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest
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roman.kudinov at novel...
Guest





PostPosted: Tue Feb 16, 2016 12:55 pm    Post subject: [Freeswitch-users] problems with bridging a call, looks like Reply with quote

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:

Quote:
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 <[url=mailto:roman.kudinov@novelapp.com]roman.kudinov@novelapp.com (roman.kudinov@novelapp.com)[/url]> wrote:
Quote:
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 ====================================
Quote:
<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 ================
Quote:
<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:
Quote:
      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
Quote:
      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
Quote:
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,G722,OPUS,PCMA"/>
in vars.xml

I tried to change
Quote:
<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
Quote:
<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


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org (consulting@freeswitch.org)
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




--

Brian West
[url=mailto:brian@freeswitch.org]brian@freeswitch.org (brian@freeswitch.org)[/url]


Twitter: @FreeSWITCH , @briankwest
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com
Got Bugs? Report them here! | Reddit: /r/freeswitch
T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest











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