With SIP/SDP the invite originator offer several codecs, but leg b is the one that set the selection. The offer by leg b, is usually based on the priority of your offer vs their highest matched codecs, so if they offer g722 as 1st priority, and you offer it as 3rd priority, they will select g722 first. Ido On Mar 6, 2016 11:50 AM, "Igor Olhovskiy" <igorolhovskiy@gmail.com> wrote:
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@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@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@32000h,G7221@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@32000h,G7221@16000h,G722,PCMU,PCMA,OPUS,SILK] outbound-codec-prefs [G7221@32000h,G7221@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
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