[Freeswitch-users] Codec negotiation. Totally confused

[Igor Olhovskiy]

Hi!
I’m getting really strange things, or I’m just missed something.
My phone is dials to freeswitch with this this line in log

2016-03-06 08:31:56.681036 [DEBUG] sofia.c:6770 Remote SDP:
v=0
o=root 1697549695 1697549695 IN IP4 <EXTERNAL IP HERE>
s=call
c=IN IP4 <EXTERNAL IP HERE>
t=0 0
m=audio 26894 RTP/AVP 9 0 8
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=nortpproxy:yes

2016-03-06 08:31:56.681036 [DEBUG] switch_core_media.c:4161 Audio Codec
Compare [G722:9:8000:20:64000:1]/[G722:9:8000:20:64000:1]
2016-03-06 08:31:56.681036 [DEBUG] switch_core_media.c:4216 Audio Codec
Compare [G722:9:8000:20:64000:1] ++++ is saved as a match
2016-03-06 08:31:56.681036 [DEBUG] switch_core_media.c:4161 Audio Codec
Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2016-03-06 08:31:56.681036 [DEBUG] switch_core_media.c:4161 Audio Codec
Compare [G722:9:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2016-03-06 08:31:56.681036 [DEBUG] switch_core_media.c:4161 Audio Codec
Compare [PCMU:0:8000:20:64000:1]/[G722:9:8000:20:64000:1]
2016-03-06 08:31:56.681036 [DEBUG] switch_core_media.c:4161 Audio Codec
Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2016-03-06 08:31:56.681036 [DEBUG] switch_core_media.c:4216 Audio Codec
Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
2016-03-06 08:31:56.681036 [DEBUG] switch_core_media.c:4161 Audio Codec
Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2016-03-06 08:31:56.681036 [DEBUG] switch_core_media.c:4161 Audio Codec
Compare [PCMA:8:8000:20:64000:1]/[G722:9:8000:20:64000:1]
2016-03-06 08:31:56.681036 [DEBUG] switch_core_media.c:4161 Audio Codec
Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2016-03-06 08:31:56.681036 [DEBUG] switch_core_media.c:4161 Audio Codec
Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2016-03-06 08:31:56.681036 [DEBUG] switch_core_media.c:4216 Audio Codec
Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
2016-03-06 08:31:56.681036 [DEBUG] switch_core_media.c:2906 Set Codec
sofia/internal/10 at consertis.securenetvox.net G722/8000 20 ms 160 samples
64000 bits 1 channels

And when switches to external profile, I see

2016-03-06 08:31:56.721010 [DEBUG] sofia_glue.c:1257
sofia/external/00972543279009 sending invite version: 1.6.6 git d2d0b32
2016-01-11 20:16:12Z 64bit
Local SDP:
v=0
o=FreeSWITCH 1457219712 1457219713 IN IP4 10.0.20.71
s=FreeSWITCH
c=IN IP4 10.0.20.71
t=0 0
m=audio 29804 RTP/AVP 9 101 13
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20
a=sendrecv


The question is - why only G722 left?
Across dialplan there is no things like inherit_codec, bypass media or
other codec-related stuff

Profiles
external
ext-sip-ip [10.0.20.71]
rtp-timeout-sec [300]
rtp-hold-timeout-sec [1800]
tls [false]
tls-only [false]
tls-bind-params [transport=tls]
tls-sip-port [5081]
tls-cert-dir [/usr/local/freeswitch/conf/ssl]
tls-passphrase []
tls-verify-date [true]
tls-verify-depth [2]
tls-verify-in-subjects []
tls-version [tlsv1]
tls-verify-policy [all]
odbc-dsn [pgsql://hostaddr=127.0.0.1 port=5432 dbname=freeswitch
user=fusionpbx password=pass options='' application_name='freeswitch']
track-calls [true]
inbound-codec-negotiation [greedy]
debug [0]
user-agent-string [FreeSWITCH]
sip-trace [no]
sip-capture [no]
rfc2833-pt [101]
sip-port [5080]
dialplan [XML]
context [public]
dtmf-duration [2000]
inbound-codec-prefs [G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,OPUS,SILK]
outbound-codec-prefs [PCMU,PCMA,GSM]
hold-music [local_stream://default]
zrtp-passthru [true]
rtp-timer-name [soft]
local-network-acl [localnet.auto]
manage-presence [false]
nonce-ttl [60]
auth-calls [false]
rtp-ip [10.0.20.71]
sip-ip [10.0.20.71]
ext-rtp-ip [10.0.20.71]

internal
tls-cert-dir [/usr/local/freeswitch/conf/ssl]
tls-passphrase []
tls-verify-date [true]
tls-verify-depth [2]
tls-verify-in-subjects []
tls-version [tlsv1]
nonce-ttl [60]
auth-calls [true]
inbound-reg-force-matching-username [true]
auth-all-packets [false]
ext-rtp-ip [10.0.20.71]
ext-sip-ip [10.0.20.71]
rtp-timeout-sec [300]
rtp-hold-timeout-sec [1800]
tls-verify-policy [all]
multiple-registrations [contact]
enable-timer [false]
dbname [share_presence]
send-presence-on-register [true]
inbound-codec-negotiation [greedy]
NDLB-force-rport [safe]
challenge-realm [auto_to]
outbound-proxy [10.0.20.70]
track-calls [true]
odbc-dsn [pgsql://hostaddr=127.0.0.1 port=5432 dbname=freeswitch
user=fusionpbx password=btgJek49 options='' application_name='freeswitch']
nat-options-ping [true]
liberal-dtmf [true]
all-reg-options-ping [true]
force-publish-expires [true]
unregister-on-options-fail [true]
user-agent-string [FreeSWITCH]
debug [0]
sip-trace [no]
sip-capture [no]
watchdog-enabled [no]
watchdog-step-timeout [30000]
watchdog-event-timeout [30000]
log-auth-failures [true]
forward-unsolicited-mwi-notify [false]
context [public]
rfc2833-pt [101]
sip-port [5060]
dialplan [XML]
dtmf-duration [2000]
inbound-codec-prefs [G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,OPUS,SILK]
outbound-codec-prefs [G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,OPUS,SILK]
rtp-timer-name [soft]
rtp-ip [10.0.20.71]
sip-ip [10.0.20.71]
hold-music [local_stream://default]
apply-nat-acl [nat.auto]
aggressive-nat-detection [true]
apply-inbound-acl [domains]
local-network-acl [localnet.auto]
record-path [/usr/local/freeswitch/recordings]
record-template
[${domain_name}/archive/${strftime(%Y)}/${strftime(%b)}/${strftime(%d)}/${uuid}.${record_ext}]
manage-presence [true]
presence-probe-on-register [true]
manage-shared-appearance [true]
tls [false]
tls-only [false]
tls-bind-params [transport=tls]
tls-sip-port [5061]

Tried with indbound-late-negotiation=false, also not helps…
Can you please, point, what is missing? Thanks

-- 
Best regards,
Igor
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