FreeSWITCH 1.6 WebRTC slow connection time
Hi guys, Since we upgraded to 1.6, it takes up to 5 seconds for media to actually flow after DTLS state becomes READY, this did not happen on 1.4. We are using sipjs and test clients on Android and iOS, the issue reproduces on all of them. Connection time used to take about 2.5-3 seconds and now it actually takes 7-8 seconds. Here are the relevant FreeSWITCH logs: 2015-06-20 16:28:36.458187 [DEBUG] sofia.c:6701 Channel sofia/internal/anonymous.jb4692@anonymous.invalid entering state [ready][200] nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nta.c:1289 agent_timer() nta: timer not set 2015-06-20 16:28:37.838233 [INFO] switch_rtp.c:3006 Changing audio DTLS state from HANDSHAKE to SETUP 2015-06-20 16:28:37.838233 [INFO] switch_rtp.c:2915 audio Fingerprint Verified. 2015-06-20 16:28:37.838233 [INFO] switch_rtp.c:3495 Activating Audio Secure RTP SEND 2015-06-20 16:28:37.838233 [DEBUG] switch_core_sqldb.c:2599 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80 2015-06-20 16:28:37.838233 [INFO] switch_rtp.c:3473 Activating Audio Secure RTP RECV 2015-06-20 16:28:37.838233 [DEBUG] switch_core_sqldb.c:2599 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80 2015-06-20 16:28:37.838233 [INFO] switch_rtp.c:2955 Changing audio DTLS state from SETUP to READY 2015-06-20 16:28:37.838233 [NOTICE] mod_dptools.c:1292 Channel [sofia/internal/anonymous.jb4692@anonymous.invalid] has been answered 2015-06-20 16:28:37.838233 [DEBUG] switch_channel.c:3759 (sofia/internal/anonymous.jb4692@anonymous.invalid) Callstate Change EARLY -> ACTIVE 2015-06-20 16:28:37.838233 [DEBUG] switch_ivr.c:195 Codec Activated L16@16000hz 1 channels 30ms 2015-06-20 16:28:37.878249 [DEBUG] switch_rtp.c:1775 rtcp_stats_init: audio ssrc[3304298062] base_seq[31262] EXECUTE sofia/internal/anonymous.jb4692@anonymous.invalid conference(1234-172.30.0.123@test) 2015-06-20 16:28:41.878188 [INFO] mod_conference.c:13067 using channel sound prefix: /usr/local/freeswitch/sounds/en/us/callie 2015-06-20 16:28:41.878188 [DEBUG] mod_conference.c:11570 Raw Codec Activation Success L16@16000hz 1 channel 30ms 2015-06-20 16:28:41.878188 [DEBUG] mod_conference.c:11617 Raw Codec Activation Success L16@16000hz 1 channel 30ms 2015-06-20 16:28:41.878188 [DEBUG] switch_core_codec.c:221 sofia/internal/anonymous.jb4692@anonymous.invalid Push codec L16:100 2015-06-20 16:28:41.878188 [DEBUG] mod_conference.c:4992 Setup timer success interval: 30 samples: 480 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '0' to 'mute' 2015-06-20 16:28:41.898176 [INFO] switch_ivr_async.c:213 Digit parser mod_conference: Setting realm to 'conf' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding 0/conf/0 callback: 0x7f429d09adb0 data: 0x7f4290166f18 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '*' to 'deaf mute' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding */conf/0 callback: 0x7f429d09adb0 data: 0x7f4290166f48 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '9' to 'energy up' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding 9/conf/0 callback: 0x7f429d09adb0 data: 0x7f4290166f78 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '8' to 'energy equ' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding 8/conf/0 callback: 0x7f429d09adb0 data: 0x7f4290166fa8 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '7' to 'energy dn' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding 7/conf/0 callback: 0x7f429d09adb0 data: 0x7f4290166fd8 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '3' to 'vol talk up' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding 3/conf/0 callback: 0x7f429d09adb0 data: 0x7f4290167008 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '2' to 'vol talk zero' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding 2/conf/0 callback: 0x7f429d09adb0 data: 0x7f4290167038 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '1' to 'vol talk dn' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding 1/conf/0 callback: 0x7f429d09adb0 data: 0x7f4290167068 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '6' to 'vol listen up' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding 6/conf/0 callback: 0x7f429d09adb0 data: 0x7f4290167098 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '5' to 'vol listen zero' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding 5/conf/0 callback: 0x7f429d09adb0 data: 0x7f42901670c8 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '4' to 'vol listen dn' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding 4/conf/0 callback: 0x7f429d09adb0 data: 0x7f42901670f8 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '#' to 'hangup' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding #/conf/0 callback: 0x7f429d09adb0 data: 0x7f4290167128 2015-06-20 16:28:41.898176 [DEBUG] mod_conference.c:6751 Setup timer soft success interval: 30 samples: 480 2015-06-20 16:28:41.918188 [DEBUG] mod_conference.c:5127 Auto recording file: /tmp/record.wav 2015-06-20 16:28:41.938180 [DEBUG] mod_conference.c:7178 Setup timer success interval: 30 samples: 480 As you can see at 2015-06-20 16:28:37.838233, the DTLS handshake completed, the call was answered and the client is thinking that media is connect but in fact only at 2015-06-20 16:28:41.938180 media opens. Any idea? Thanks, Adam
can you confirm if the behavior is the same using verto? please open a bug at https://FreeSWITCH.org/jira on this issue. On Saturday, June 20, 2015, Adam Ben-Ayoun <adam.ben.ayoun1@gmail.com> wrote:
Hi guys,
Since we upgraded to 1.6, it takes up to 5 seconds for media to actually flow after DTLS state becomes READY, this did not happen on 1.4. We are using sipjs and test clients on Android and iOS, the issue reproduces on all of them. Connection time used to take about 2.5-3 seconds and now it actually takes 7-8 seconds.
Here are the relevant FreeSWITCH logs:
2015-06-20 16:28:36.458187 [DEBUG] sofia.c:6701 Channel sofia/internal/anonymous.jb4692@anonymous.invalid entering state [ready][200] nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nta.c:1289 agent_timer() nta: timer not set 2015-06-20 16:28:37.838233 [INFO] switch_rtp.c:3006 Changing audio DTLS state from HANDSHAKE to SETUP 2015-06-20 16:28:37.838233 [INFO] switch_rtp.c:2915 audio Fingerprint Verified. 2015-06-20 16:28:37.838233 [INFO] switch_rtp.c:3495 Activating Audio Secure RTP SEND 2015-06-20 16:28:37.838233 [DEBUG] switch_core_sqldb.c:2599 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80 2015-06-20 16:28:37.838233 [INFO] switch_rtp.c:3473 Activating Audio Secure RTP RECV 2015-06-20 16:28:37.838233 [DEBUG] switch_core_sqldb.c:2599 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80 2015-06-20 16:28:37.838233 [INFO] switch_rtp.c:2955 Changing audio DTLS state from SETUP to READY 2015-06-20 16:28:37.838233 [NOTICE] mod_dptools.c:1292 Channel [sofia/internal/anonymous.jb4692@anonymous.invalid] has been answered 2015-06-20 16:28:37.838233 [DEBUG] switch_channel.c:3759 (sofia/internal/anonymous.jb4692@anonymous.invalid) Callstate Change EARLY -> ACTIVE 2015-06-20 16:28:37.838233 [DEBUG] switch_ivr.c:195 Codec Activated L16@16000hz 1 channels 30ms 2015-06-20 16:28:37.878249 [DEBUG] switch_rtp.c:1775 rtcp_stats_init: audio ssrc[3304298062] base_seq[31262] EXECUTE sofia/internal/anonymous.jb4692@anonymous.invalid conference(1234-172.30.0.123@test) 2015-06-20 16:28:41.878188 [INFO] mod_conference.c:13067 using channel sound prefix: /usr/local/freeswitch/sounds/en/us/callie 2015-06-20 16:28:41.878188 [DEBUG] mod_conference.c:11570 Raw Codec Activation Success L16@16000hz 1 channel 30ms 2015-06-20 16:28:41.878188 [DEBUG] mod_conference.c:11617 Raw Codec Activation Success L16@16000hz 1 channel 30ms 2015-06-20 16:28:41.878188 [DEBUG] switch_core_codec.c:221 sofia/internal/anonymous.jb4692@anonymous.invalid Push codec L16:100 2015-06-20 16:28:41.878188 [DEBUG] mod_conference.c:4992 Setup timer success interval: 30 samples: 480 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '0' to 'mute' 2015-06-20 16:28:41.898176 [INFO] switch_ivr_async.c:213 Digit parser mod_conference: Setting realm to 'conf' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding 0/conf/0 callback: 0x7f429d09adb0 data: 0x7f4290166f18 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '*' to 'deaf mute' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding */conf/0 callback: 0x7f429d09adb0 data: 0x7f4290166f48 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '9' to 'energy up' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding 9/conf/0 callback: 0x7f429d09adb0 data: 0x7f4290166f78 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '8' to 'energy equ' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding 8/conf/0 callback: 0x7f429d09adb0 data: 0x7f4290166fa8 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '7' to 'energy dn' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding 7/conf/0 callback: 0x7f429d09adb0 data: 0x7f4290166fd8 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '3' to 'vol talk up' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding 3/conf/0 callback: 0x7f429d09adb0 data: 0x7f4290167008 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '2' to 'vol talk zero' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding 2/conf/0 callback: 0x7f429d09adb0 data: 0x7f4290167038 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '1' to 'vol talk dn' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding 1/conf/0 callback: 0x7f429d09adb0 data: 0x7f4290167068 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '6' to 'vol listen up' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding 6/conf/0 callback: 0x7f429d09adb0 data: 0x7f4290167098 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '5' to 'vol listen zero' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding 5/conf/0 callback: 0x7f429d09adb0 data: 0x7f42901670c8 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '4' to 'vol listen dn' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding 4/conf/0 callback: 0x7f429d09adb0 data: 0x7f42901670f8 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '#' to 'hangup' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding #/conf/0 callback: 0x7f429d09adb0 data: 0x7f4290167128 2015-06-20 16:28:41.898176 [DEBUG] mod_conference.c:6751 Setup timer soft success interval: 30 samples: 480 2015-06-20 16:28:41.918188 [DEBUG] mod_conference.c:5127 Auto recording file: /tmp/record.wav 2015-06-20 16:28:41.938180 [DEBUG] mod_conference.c:7178 Setup timer success interval: 30 samples: 480
As you can see at 2015-06-20 16:28:37.838233, the DTLS handshake completed, the call was answered and the client is thinking that media is connect but in fact only at 2015-06-20 16:28:41.938180 media opens. Any idea?
Thanks, Adam
Yes. I will open a ticket. On 20 June 2015 at 22:58, Michael Jerris <mike@jerris.com> wrote:
can you confirm if the behavior is the same using verto? please open a bug at https://FreeSWITCH.org/jira on this issue.
On Saturday, June 20, 2015, Adam Ben-Ayoun <adam.ben.ayoun1@gmail.com> wrote:
Hi guys,
Since we upgraded to 1.6, it takes up to 5 seconds for media to actually flow after DTLS state becomes READY, this did not happen on 1.4. We are using sipjs and test clients on Android and iOS, the issue reproduces on all of them. Connection time used to take about 2.5-3 seconds and now it actually takes 7-8 seconds.
Here are the relevant FreeSWITCH logs:
2015-06-20 16:28:36.458187 [DEBUG] sofia.c:6701 Channel sofia/internal/anonymous.jb4692@anonymous.invalid entering state [ready][200] nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nta.c:1289 agent_timer() nta: timer not set 2015-06-20 16:28:37.838233 [INFO] switch_rtp.c:3006 Changing audio DTLS state from HANDSHAKE to SETUP 2015-06-20 16:28:37.838233 [INFO] switch_rtp.c:2915 audio Fingerprint Verified. 2015-06-20 16:28:37.838233 [INFO] switch_rtp.c:3495 Activating Audio Secure RTP SEND 2015-06-20 16:28:37.838233 [DEBUG] switch_core_sqldb.c:2599 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80 2015-06-20 16:28:37.838233 [INFO] switch_rtp.c:3473 Activating Audio Secure RTP RECV 2015-06-20 16:28:37.838233 [DEBUG] switch_core_sqldb.c:2599 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80 2015-06-20 16:28:37.838233 [INFO] switch_rtp.c:2955 Changing audio DTLS state from SETUP to READY 2015-06-20 16:28:37.838233 [NOTICE] mod_dptools.c:1292 Channel [sofia/internal/anonymous.jb4692@anonymous.invalid] has been answered 2015-06-20 16:28:37.838233 [DEBUG] switch_channel.c:3759 (sofia/internal/anonymous.jb4692@anonymous.invalid) Callstate Change EARLY -> ACTIVE 2015-06-20 16:28:37.838233 [DEBUG] switch_ivr.c:195 Codec Activated L16@16000hz 1 channels 30ms 2015-06-20 16:28:37.878249 [DEBUG] switch_rtp.c:1775 rtcp_stats_init: audio ssrc[3304298062] base_seq[31262] EXECUTE sofia/internal/anonymous.jb4692@anonymous.invalid conference(1234-172.30.0.123@test) 2015-06-20 16:28:41.878188 [INFO] mod_conference.c:13067 using channel sound prefix: /usr/local/freeswitch/sounds/en/us/callie 2015-06-20 16:28:41.878188 [DEBUG] mod_conference.c:11570 Raw Codec Activation Success L16@16000hz 1 channel 30ms 2015-06-20 16:28:41.878188 [DEBUG] mod_conference.c:11617 Raw Codec Activation Success L16@16000hz 1 channel 30ms 2015-06-20 16:28:41.878188 [DEBUG] switch_core_codec.c:221 sofia/internal/anonymous.jb4692@anonymous.invalid Push codec L16:100 2015-06-20 16:28:41.878188 [DEBUG] mod_conference.c:4992 Setup timer success interval: 30 samples: 480 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '0' to 'mute' 2015-06-20 16:28:41.898176 [INFO] switch_ivr_async.c:213 Digit parser mod_conference: Setting realm to 'conf' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding 0/conf/0 callback: 0x7f429d09adb0 data: 0x7f4290166f18 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '*' to 'deaf mute' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding */conf/0 callback: 0x7f429d09adb0 data: 0x7f4290166f48 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '9' to 'energy up' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding 9/conf/0 callback: 0x7f429d09adb0 data: 0x7f4290166f78 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '8' to 'energy equ' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding 8/conf/0 callback: 0x7f429d09adb0 data: 0x7f4290166fa8 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '7' to 'energy dn' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding 7/conf/0 callback: 0x7f429d09adb0 data: 0x7f4290166fd8 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '3' to 'vol talk up' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding 3/conf/0 callback: 0x7f429d09adb0 data: 0x7f4290167008 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '2' to 'vol talk zero' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding 2/conf/0 callback: 0x7f429d09adb0 data: 0x7f4290167038 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '1' to 'vol talk dn' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding 1/conf/0 callback: 0x7f429d09adb0 data: 0x7f4290167068 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '6' to 'vol listen up' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding 6/conf/0 callback: 0x7f429d09adb0 data: 0x7f4290167098 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '5' to 'vol listen zero' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding 5/conf/0 callback: 0x7f429d09adb0 data: 0x7f42901670c8 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '4' to 'vol listen dn' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding 4/conf/0 callback: 0x7f429d09adb0 data: 0x7f42901670f8 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '#' to 'hangup' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding #/conf/0 callback: 0x7f429d09adb0 data: 0x7f4290167128 2015-06-20 16:28:41.898176 [DEBUG] mod_conference.c:6751 Setup timer soft success interval: 30 samples: 480 2015-06-20 16:28:41.918188 [DEBUG] mod_conference.c:5127 Auto recording file: /tmp/record.wav 2015-06-20 16:28:41.938180 [DEBUG] mod_conference.c:7178 Setup timer success interval: 30 samples: 480
As you can see at 2015-06-20 16:28:37.838233, the DTLS handshake completed, the call was answered and the client is thinking that media is connect but in fact only at 2015-06-20 16:28:41.938180 media opens. Any idea?
Thanks, Adam
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The ticket is at https://freeswitch.org/jira/browse/FS-7703 for reference. On 22 June 2015 at 10:04, Adam Ben-Ayoun <adam.ben.ayoun1@gmail.com> wrote:
Yes. I will open a ticket.
On 20 June 2015 at 22:58, Michael Jerris <mike@jerris.com> wrote:
can you confirm if the behavior is the same using verto? please open a bug at https://FreeSWITCH.org/jira on this issue.
On Saturday, June 20, 2015, Adam Ben-Ayoun <adam.ben.ayoun1@gmail.com> wrote:
Hi guys,
Since we upgraded to 1.6, it takes up to 5 seconds for media to actually flow after DTLS state becomes READY, this did not happen on 1.4. We are using sipjs and test clients on Android and iOS, the issue reproduces on all of them. Connection time used to take about 2.5-3 seconds and now it actually takes 7-8 seconds.
Here are the relevant FreeSWITCH logs:
2015-06-20 16:28:36.458187 [DEBUG] sofia.c:6701 Channel sofia/internal/anonymous.jb4692@anonymous.invalid entering state [ready][200] nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nta.c:1289 agent_timer() nta: timer not set 2015-06-20 16:28:37.838233 [INFO] switch_rtp.c:3006 Changing audio DTLS state from HANDSHAKE to SETUP 2015-06-20 16:28:37.838233 [INFO] switch_rtp.c:2915 audio Fingerprint Verified. 2015-06-20 16:28:37.838233 [INFO] switch_rtp.c:3495 Activating Audio Secure RTP SEND 2015-06-20 16:28:37.838233 [DEBUG] switch_core_sqldb.c:2599 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80 2015-06-20 16:28:37.838233 [INFO] switch_rtp.c:3473 Activating Audio Secure RTP RECV 2015-06-20 16:28:37.838233 [DEBUG] switch_core_sqldb.c:2599 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80 2015-06-20 16:28:37.838233 [INFO] switch_rtp.c:2955 Changing audio DTLS state from SETUP to READY 2015-06-20 16:28:37.838233 [NOTICE] mod_dptools.c:1292 Channel [sofia/internal/anonymous.jb4692@anonymous.invalid] has been answered 2015-06-20 16:28:37.838233 [DEBUG] switch_channel.c:3759 (sofia/internal/anonymous.jb4692@anonymous.invalid) Callstate Change EARLY -> ACTIVE 2015-06-20 16:28:37.838233 [DEBUG] switch_ivr.c:195 Codec Activated L16@16000hz 1 channels 30ms 2015-06-20 16:28:37.878249 [DEBUG] switch_rtp.c:1775 rtcp_stats_init: audio ssrc[3304298062] base_seq[31262] EXECUTE sofia/internal/anonymous.jb4692@anonymous.invalid conference(1234-172.30.0.123@test) 2015-06-20 16:28:41.878188 [INFO] mod_conference.c:13067 using channel sound prefix: /usr/local/freeswitch/sounds/en/us/callie 2015-06-20 16:28:41.878188 [DEBUG] mod_conference.c:11570 Raw Codec Activation Success L16@16000hz 1 channel 30ms 2015-06-20 16:28:41.878188 [DEBUG] mod_conference.c:11617 Raw Codec Activation Success L16@16000hz 1 channel 30ms 2015-06-20 16:28:41.878188 [DEBUG] switch_core_codec.c:221 sofia/internal/anonymous.jb4692@anonymous.invalid Push codec L16:100 2015-06-20 16:28:41.878188 [DEBUG] mod_conference.c:4992 Setup timer success interval: 30 samples: 480 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '0' to 'mute' 2015-06-20 16:28:41.898176 [INFO] switch_ivr_async.c:213 Digit parser mod_conference: Setting realm to 'conf' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding 0/conf/0 callback: 0x7f429d09adb0 data: 0x7f4290166f18 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '*' to 'deaf mute' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding */conf/0 callback: 0x7f429d09adb0 data: 0x7f4290166f48 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '9' to 'energy up' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding 9/conf/0 callback: 0x7f429d09adb0 data: 0x7f4290166f78 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '8' to 'energy equ' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding 8/conf/0 callback: 0x7f429d09adb0 data: 0x7f4290166fa8 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '7' to 'energy dn' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding 7/conf/0 callback: 0x7f429d09adb0 data: 0x7f4290166fd8 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '3' to 'vol talk up' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding 3/conf/0 callback: 0x7f429d09adb0 data: 0x7f4290167008 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '2' to 'vol talk zero' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding 2/conf/0 callback: 0x7f429d09adb0 data: 0x7f4290167038 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '1' to 'vol talk dn' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding 1/conf/0 callback: 0x7f429d09adb0 data: 0x7f4290167068 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '6' to 'vol listen up' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding 6/conf/0 callback: 0x7f429d09adb0 data: 0x7f4290167098 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '5' to 'vol listen zero' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding 5/conf/0 callback: 0x7f429d09adb0 data: 0x7f42901670c8 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '4' to 'vol listen dn' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding 4/conf/0 callback: 0x7f429d09adb0 data: 0x7f42901670f8 2015-06-20 16:28:41.898176 [INFO] mod_conference.c:13676 sofia/internal/anonymous.jb4692@anonymous.invalid binding '#' to 'hangup' 2015-06-20 16:28:41.898176 [DEBUG] switch_ivr_async.c:322 Digit parser mod_conference: binding #/conf/0 callback: 0x7f429d09adb0 data: 0x7f4290167128 2015-06-20 16:28:41.898176 [DEBUG] mod_conference.c:6751 Setup timer soft success interval: 30 samples: 480 2015-06-20 16:28:41.918188 [DEBUG] mod_conference.c:5127 Auto recording file: /tmp/record.wav 2015-06-20 16:28:41.938180 [DEBUG] mod_conference.c:7178 Setup timer success interval: 30 samples: 480
As you can see at 2015-06-20 16:28:37.838233, the DTLS handshake completed, the call was answered and the client is thinking that media is connect but in fact only at 2015-06-20 16:28:41.938180 media opens. Any idea?
Thanks, Adam
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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Hi all, I try to record calls with video in default dialplan. I’m use mod_av for recording video <extension name="Local_Extension"> <condition field="destination_number" expression="^(10[01][0-9]|^(onops\d{1,6})$)$"> <action application="export" data="dialed_extension=$1"/> <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="bind_meta_app" data="4 b s execute_extension::att_xfer XML features"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=30"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}"/> <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> <action application="set" data="record_concat_video=true"/> <action application="record_session" data="/usr/local/freeswitch/recordings/${strftime(%Y%m%d_%H%M%S)}_${caller_id_number}_${destination_number}.mp4"/> <action application="bridge" data="{ignore_early_media=true}user/${dialed_extension}@${domain_name}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> </condition> </extension> And it works somehow strange. Video recording is excellent. But the sound exactly to the middle. After the sound disappears and only the video continues. I have the feeling that this module is somehow not properly or generates a file completes. Any ideas? -- S pozdravem, Ing.Denis Jakovlev mob.tel. 775-415-382
what operating system are you using?
On Jun 22, 2015, at 5:42 AM, Denis Jakovlev <yadenis@seznam.cz> wrote:
Hi all,
I try to record calls with video in default dialplan. I’m use mod_av for recording video
<extension name="Local_Extension"> <condition field="destination_number" expression="^(10[01][0-9]|^(onops\d{1,6})$)$"> <action application="export" data="dialed_extension=$1"/> <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="bind_meta_app" data="4 b s execute_extension::att_xfer XML features"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=30"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}"/> <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
<action application="set" data="record_concat_video=true"/> <action application="record_session" data="/usr/local/freeswitch/recordings/${strftime(%Y%m%d_%H%M%S)}_${caller_id_number}_${destination_number}.mp4"/>
<action application="bridge" data="{ignore_early_media=true}user/${dialed_extension}@${domain_name}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> </condition> </extension>
And it works somehow strange. Video recording is excellent. But the sound exactly to the middle. After the sound disappears and only the video continues. I have the feeling that this module is somehow not properly or generates a file completes.
Any ideas?
-- S pozdravem, Ing.Denis Jakovlev mob.tel. 775-415-382
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Dobrý den. Debian 8 -- S pozdravem, Ing.Denis Jakovlev mob.tel. 775-415-382
On 22. 6. 2015, at 20:09, Michael Jerris <mike@jerris.com> wrote:
what operating system are you using?
On Jun 22, 2015, at 5:42 AM, Denis Jakovlev <yadenis@seznam.cz <mailto:yadenis@seznam.cz>> wrote:
Hi all,
I try to record calls with video in default dialplan. I’m use mod_av for recording video
<extension name="Local_Extension"> <condition field="destination_number" expression="^(10[01][0-9]|^(onops\d{1,6})$)$"> <action application="export" data="dialed_extension=$1"/> <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="bind_meta_app" data="4 b s execute_extension::att_xfer XML features"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=30"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}"/> <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
<action application="set" data="record_concat_video=true"/> <action application="record_session" data="/usr/local/freeswitch/recordings/${strftime(%Y%m%d_%H%M%S)}_${caller_id_number}_${destination_number}.mp4"/>
<action application="bridge" data="{ignore_early_media=true}user/${dialed_extension}@${domain_name}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> </condition> </extension>
And it works somehow strange. Video recording is excellent. But the sound exactly to the middle. After the sound disappears and only the video continues. I have the feeling that this module is somehow not properly or generates a file completes.
Any ideas?
-- S pozdravem, Ing.Denis Jakovlev mob.tel. 775-415-382
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
dpkg -l | grep libav | grep extra On Tuesday, June 23, 2015, Denis Jakovlev <yadenis@seznam.cz> wrote:
Dobrý den.
Debian 8
*-- * *S pozdravem,* *Ing.Denis Jakovlev * *mob.tel <http://mob.tel>. 775-415-382*
On 22. 6. 2015, at 20:09, Michael Jerris <mike@jerris.com <javascript:_e(%7B%7D,'cvml','mike@jerris.com');>> wrote:
what operating system are you using?
On Jun 22, 2015, at 5:42 AM, Denis Jakovlev <yadenis@seznam.cz <javascript:_e(%7B%7D,'cvml','yadenis@seznam.cz');>> wrote:
Hi all,
I try to record calls with video in default dialplan. I’m use mod_av for recording video
<extension name="Local_Extension"> <condition field="destination_number" expression="^(10[01][0-9]|^(onops\d{1,6})$)$"> <action application="export" data="dialed_extension=$1"/> <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="bind_meta_app" data="4 b s execute_extension::att_xfer XML features"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=30"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}"/> <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
*<action application="set" data="record_concat_video=true"/> <action application="record_session" data="/usr/local/freeswitch/recordings/${strftime(%Y%m%d_%H%M%S)}_${caller_id_number}_${destination_number}.mp4"/> *<action application="bridge" data="{ignore_early_media=true}user/${dialed_extension}@${domain_name}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> </condition> </extension>
And it works somehow strange. Video recording is excellent. But the sound exactly to the middle. After the sound disappears and only the video continues. I have the feeling that this module is somehow not properly or generates a file completes.
Any ideas?
*-- S pozdravem, Ing.Denis Jakovlev mob.tel <http://mob.tel>. 775-415-382 *
*_________________________________________________________________________Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <javascript:_e(%7B%7D,'cvml','consulting@freeswitch.org');>http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com>Official FreeSWITCH Siteshttp://www.freeswitch.org <http://www.freeswitch.org>http://confluence.freeswitch.org <http://confluence.freeswitch.org>http://www.cluecon.com <http://www.cluecon.com>FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.org <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users@lists.freeswitch.org');>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>http://www.freeswitch.org <http://www.freeswitch.org>*
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <javascript:_e(%7B%7D,'cvml','consulting@freeswitch.org');> http://www.freeswitchsolutions.com
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FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users@lists.freeswitch.org');> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Dobrý den, dpkg -l | grep libav | grep extra ii libavcodec-extra 6:11.4-1~deb8u1 all Libav codec library (additional codecs meta-package) ii libavcodec-extra-56:amd64 6:11.4-1~deb8u1 amd64 Libav codec library (additional codecs) -- S pozdravem, Ing.Denis Jakovlev mob.tel. 775-415-382 úterý 23. června 2015, 17:15:34, napsal jste: dpkg -l | grep libav | grep extra On Tuesday, June 23, 2015, Denis Jakovlev <yadenis@seznam.cz> wrote: Dobrý den. Debian 8 -- S pozdravem, Ing.Denis Jakovlev mob.tel. 775-415-382 On 22. 6. 2015, at 20:09, Michael Jerris <mike@jerris.com> wrote: what operating system are you using? On Jun 22, 2015, at 5:42 AM, Denis Jakovlev <yadenis@seznam.cz> wrote: Hi all, I try to record calls with video in default dialplan. I’m use mod_av for recording video <extension name="Local_Extension"> <condition field="destination_number" expression="^(10[01][0-9]|^(onops\d{1,6})$)$"> <action application="export" data="dialed_extension=$1"/> <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="bind_meta_app" data="4 b s execute_extension::att_xfer XML features"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=30"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}"/> <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> <action application="set" data="record_concat_video=true"/> <action application="record_session" data="/usr/local/freeswitch/recordings/${strftime(%Y%m%d_%H%M%S)}_${caller_id_number}_${destination_number}.mp4"/> <action application="bridge" data="{ignore_early_media=true}user/${dialed_extension}@${domain_name}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> </condition> </extension> And it works somehow strange. Video recording is excellent. But the sound exactly to the middle. After the sound disappears and only the video continues. I have the feeling that this module is somehow not properly or generates a file completes. Any ideas? -- S pozdravem, Ing.Denis Jakovlev mob.tel. 775-415-382 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
I've posted three working test cases in the testing config on your JIRA, the record session bits work fine, can you test the three test cases I posted? https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/conf/testin... On Mon, Jun 22, 2015 at 4:42 AM, Denis Jakovlev <yadenis@seznam.cz> wrote:
Hi all,
I try to record calls with video in default dialplan. I’m use mod_av for recording video
<extension name="Local_Extension"> <condition field="destination_number" expression="^(10[01][0-9]|^(onops\d{1,6})$)$"> <action application="export" data="dialed_extension=$1"/> <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="bind_meta_app" data="4 b s execute_extension::att_xfer XML features"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=30"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}"/> <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
*<action application="set" data="record_concat_video=true"/> <action application="record_session" data="/usr/local/freeswitch/recordings/${strftime(%Y%m%d_%H%M%S)}_${caller_id_number}_${destination_number}.mp4"/> *<action application="bridge" data="{ignore_early_media=true}user/${dialed_extension}@${domain_name}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> </condition> </extension>
And it works somehow strange. Video recording is excellent. But the sound exactly to the middle. After the sound disappears and only the video continues. I have the feeling that this module is somehow not properly or generates a file completes.
Any ideas?
*-- S pozdravem, Ing.Denis Jakovlev mob.tel <http://mob.tel>. 775-415-382 *
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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participants (4)
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Adam Ben-Ayoun
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Brian West
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Denis Jakovlev
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Michael Jerris