OpenZAP codec Question: Why L16@8000 codec for incoming calls
Hello, today I found in FS logfile lines like this: 2009-02-23 15:27:12 [DEBUG] switch_ivr_originate.c:1605 switch_ivr_originate() Raw Codec Activation Success L16@8000hz 1 channel 20ms It looks like L16 codec is used for incoming calls: 2009-02-23 15:27:03 [DEBUG] switch_core_session.c:523 switch_core_session_perform_receive_message() Send signal OpenZAP/1:18/2799 [BREAK] 2009-02-23 15:27:03 [NOTICE] switch_ivr_originate.c:1588 switch_ivr_originate() Pre-Answer OpenZAP/1:18/2799! 2009-02-23 15:27:03 [DEBUG] switch_ivr_originate.c:1605 switch_ivr_originate() Raw Codec Activation Success L16@8000hz 1 channel 20ms 2009-02-23 15:27:03 [DEBUG] switch_ivr_originate.c:1664 switch_ivr_originate() Play Ringback Tone [%(1000, 4000, 425.0, 0)] 2009-02-23 15:27:03 [DEBUG] sofia.c:2725 sofia_handle_sip_i_state() Channel sofia/internal/sip:2799@85.16.245.254:5060;user=phone;transport=udp entering state [proceeding] 2009-02-23 15:27:03 [NOTICE] sofia.c:2779 sofia_handle_sip_i_state() Ring-Ready sofia/internal/sip:2799@85.16.245.254:5060;user=phone;transport=udp! 2009-02-23 15:27:03 [DEBUG] switch_core_io.c:652 switch_core_session_write_frame() OpenZAP/1:18/2799 receive message [TRANSCODING_NECESSARY] 2009-02-23 15:27:07 [DEBUG] Span:1 Q.931() Timer 0 of call 6 (CRV: 61, State: 0) timed out 2009-02-23 15:27:12 [DEBUG] sofia.c:2725 sofia_handle_sip_i_state() Channel sofia/internal/sip:2799@85.16.245.254:5060;user=phone;transport=udp entering state [ready] 2009-02-23 15:27:12 [DEBUG] sofia.c:2729 sofia_handle_sip_i_state() Remote SDP: v=0^M o=2799 121183017 121183017 IN IP4 85.16.245.254^M s=ATA186 Call^M c=IN IP4 85.16.245.254^M t=0 0^M m=audio 16384 RTP/AVP 8 101^M a=rtpmap:8 PCMA/8000/1^M a=rtpmap:101 telephone-event/8000^M a=fmtp:101 0-15^M 2009-02-23 15:27:12 [DEBUG] sofia_glue.c:2549 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMA:8:8000:0]/[PCMA:8:8000:20] 2009-02-23 15:27:12 [DEBUG] sofia_glue.c:1684 sofia_glue_tech_set_codec() Set Codec sofia/internal/sip:2799@85.16.245.254:5060;user=phone;transport=udp PCMA/8000 20 ms 160 samples The audio codec compare function finds slightly different codecs for A and B party. The dialplan for incoming calls via openzap is this. I set the codec to use in extensions "bridge" line: <extension name="fp_Local_Extension"> <condition field="destination_number" expression="(491[0-9]|492[0-8])$"> <action application="ring_ready"/> <action application="set" data="ringback=${de-ring}"/> <action application="export" data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name} var sip_secure_media)}"/> <action application="bridge" data="{absolute_codec_string=PCMA}user/$1@$${domain}"/> </condition> </extension> In my vars.xml config I have these codecs configured: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G722,PCMA"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G722,PCMA"/> So where can I disable the L16 codec, or why is a transcoding necessary? regards Helmut
On Feb 23, 2009, at 9:44 AM, Helmut Kuper wrote:
Hello,
today I found in FS logfile lines like this:
2009-02-23 15:27:12 [DEBUG] switch_ivr_originate.c:1605 switch_ivr_originate() Raw Codec Activation Success L16@8000hz 1 channel 20ms
It looks like L16 codec is used for incoming calls:
2009-02-23 15:27:03 [DEBUG] switch_core_session.c:523 switch_core_session_perform_receive_message() Send signal OpenZAP/1:18/2799 [BREAK] 2009-02-23 15:27:03 [NOTICE] switch_ivr_originate.c:1588 switch_ivr_originate() Pre-Answer OpenZAP/1:18/2799! 2009-02-23 15:27:03 [DEBUG] switch_ivr_originate.c:1605 switch_ivr_originate() Raw Codec Activation Success L16@8000hz 1 channel 20ms 2009-02-23 15:27:03 [DEBUG] switch_ivr_originate.c:1664 switch_ivr_originate() Play Ringback Tone [%(1000, 4000, 425.0, 0)] 2009-02-23 15:27:03 [DEBUG] sofia.c:2725 sofia_handle_sip_i_state() Channel sofia/internal/sip:2799@85.16.245.254:5060;user=phone;transport=udp entering state [proceeding] 2009-02-23 15:27:03 [NOTICE] sofia.c:2779 sofia_handle_sip_i_state() Ring-Ready sofia/internal/sip:2799@85.16.245.254:5060;user=phone;transport=udp! 2009-02-23 15:27:03 [DEBUG] switch_core_io.c:652 switch_core_session_write_frame() OpenZAP/1:18/2799 receive message [TRANSCODING_NECESSARY] 2009-02-23 15:27:07 [DEBUG] Span:1 Q.931() Timer 0 of call 6 (CRV: 61, State: 0) timed out 2009-02-23 15:27:12 [DEBUG] sofia.c:2725 sofia_handle_sip_i_state() Channel sofia/internal/sip:2799@85.16.245.254:5060;user=phone;transport=udp entering state [ready] 2009-02-23 15:27:12 [DEBUG] sofia.c:2729 sofia_handle_sip_i_state() Remote SDP: v=0^M o=2799 121183017 121183017 IN IP4 85.16.245.254^M s=ATA186 Call^M c=IN IP4 85.16.245.254^M t=0 0^M m=audio 16384 RTP/AVP 8 101^M a=rtpmap:8 PCMA/8000/1^M a=rtpmap:101 telephone-event/8000^M a=fmtp:101 0-15^M
2009-02-23 15:27:12 [DEBUG] sofia_glue.c:2549 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMA:8:8000:0]/[PCMA:8:8000:20] 2009-02-23 15:27:12 [DEBUG] sofia_glue.c:1684 sofia_glue_tech_set_codec() Set Codec sofia/internal/sip:2799@85.16.245.254:5060;user=phone;transport=udp PCMA/8000 20 ms 160 samples
The audio codec compare function finds slightly different codecs for A and B party.
The dialplan for incoming calls via openzap is this. I set the codec to use in extensions "bridge" line:
<extension name="fp_Local_Extension"> <condition field="destination_number" expression="(491[0-9]|492[0-8])$"> <action application="ring_ready"/> <action application="set" data="ringback=${de-ring}"/> <action application="export" data="nolocal:sip_secure_media=${user_data(${dialed_extension}@$ {domain_name} var sip_secure_media)}"/> <action application="bridge" data="{absolute_codec_string=PCMA}user/$1@$${domain}"/> </condition> </extension>
In my vars.xml config I have these codecs configured:
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G722,PCMA"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G722,PCMA"/>
So where can I disable the L16 codec, or why is a transcoding necessary?
Your playing a tone, we need to encode that tone into the codec of the channel. You could make it stop transcoding by not providing ringback but we are still doing some transcoding for the tone detection in openzap that you won't see via log messages. Why is this transcoding a problem? Mike
Hi Mike, thx. Today we had some failing test fax sessions (g711/PCMA) and my first thought was that it could be caused by FS during transcoding. So I looked into FS logfile and found those hints about transcoding. But ringback shouldn't be the problem. Fax path was from FAX device (source) through a voip cpe, through a SBC through a SS7 PSTN device into PSTN. Then from PSTN through AVAYA PBX trough sangoma A104d into freeswitch. From there RTP goes to a Cisco ATA to be converted to TDM and consumed by a FAX device (Target). So we have a lot of points to look at ... regrads Helmut
Your playing a tone, we need to encode that tone into the codec of the channel. You could make it stop transcoding by not providing ringback but we are still doing some transcoding for the tone detection in openzap that you won't see via log messages. Why is this transcoding a problem?
Mike
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Hi Mike, thx. Today we had some failing test fax sessions (g711/PCMA) and my first thought was that it could be caused by FS during transcoding. So I looked into FS logfile and found those hints about transcoding. But ringback shouldn't be the problem. Fax path was from FAX device (source) through a voip cpe, through a SBC through a SS7 PSTN device into PSTN. Then from PSTN through AVAYA PBX trough sangoma A104d into freeswitch. From there RTP goes to a Cisco ATA to be converted to TDM and consumed by a FAX device (Target). So we have a lot of points to look at ... regrads Helmut
Your playing a tone, we need to encode that tone into the codec of the channel. You could make it stop transcoding by not providing ringback but we are still doing some transcoding for the tone detection in openzap that you won't see via log messages. Why is this transcoding a problem?
Mike
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participants (2)
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Helmut Kuper
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Michael Jerris