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[helmut.kuper at ewetel.de]

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



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[mike at jerris.com]

On Feb 23, 2009, at 9:44 AM, Helmut Kuper wrote:

[helmut.kuper at ewetel.de]

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

[helmut.kuper at ewetel.de]

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