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

[Helmut Kuper]

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?

regards
Helmut