[Freeswitch-users] Codec negotiation. Totally confused

Volodymyr Fedorov lexxua at gmail.com
Sun Mar 6 23:45:28 MSK 2016


Hi Igor,
If you want to do transcoding try to set*
media_mix_inbound_outbound_codecs=true.* This is described in detail here :
https://freeswitch.org/confluence/plugins/servlet/mobile#content/view/2883752
.
On Mar 6, 2016 11:28 AM, "Igor Olhovskiy" <igorolhovskiy at gmail.com> wrote:

> Working only when I’m setting
> export nolocal:absolute_codec_string=${outbound_codec_prefs}
>
>
> 2016-03-06 11:47 GMT+02:00 Igor Olhovskiy <igorolhovskiy at gmail.com>:
>
>> Main question - why it’s ignores outbound-codec-prefs on external
>> profile and use G722 as a first avail codec in list?
>>
>> 2016-03-06 9:51 GMT+02:00 Igor Olhovskiy <igorolhovskiy at gmail.com>:
>>
>>> 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
>>>
>>
>>
>>
>> --
>> Best regards,
>> Igor
>>
>
>
>
> --
> Best regards,
> Igor
>
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