[Freeswitch-users] Codec negotiation. Totally confused

Igor Olhovskiy igorolhovskiy at gmail.com
Sun Mar 6 13:25:18 MSK 2016


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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