[Freeswitch-users] OpenZAP codec Question: Why L16 at 8000 codec for incoming calls

[Michael Jerris]

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 at 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 at 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 at 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 at 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 at 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 at 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