[asterisk-bugs] [JIRA] (ASTERISK-26544) Delay in DTLS handshake causes audio setup delay
Asterisk Team (JIRA)
noreply at issues.asterisk.org
Tue Nov 1 11:28:10 CDT 2016
[ https://issues.asterisk.org/jira/browse/ASTERISK-26544?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=233251#comment-233251 ]
Asterisk Team commented on ASTERISK-26544:
------------------------------------------
Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.
A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.
Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].
> Delay in DTLS handshake causes audio setup delay
> ------------------------------------------------
>
> Key: ASTERISK-26544
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-26544
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Resources/res_rtp_asterisk
> Affects Versions: 13.6.0, 13.9.0, 13.12.0, 13.11.2, 13.12.1, 14.0.0, 14.1.1
> Environment: Amazon Linux: 4.4.23-31.54.amzn1.x86_64 #1 SMP x86_64 GNU/Linux
> OpenSSL: 1.0.2j
> LibSRTP: 1.5.4
> SIPml5: e3152e1edf116b651de379b3cc971bf699787c26 (Fri Mar 4 09:47:48 2016 +0100)
> Chrome: 54.0.2840.71 (64-bit)
> FireFox: 49.0.2
> Opera: 41.0
> Online JSSip Demo at: https://tryit.jssip.net/
> Amazon EC2 instance
> Reporter: Marcelo Gornstein
>
> Hello,
> It seems that there is a delay in the audio setup when using WebRTC with latest Asterisk versions and latest browser versions (described in the Environment section).
> Sometimes there is no delay, but most of the time the delay goes between 1 second to a couple of minutes.
> This seems to be related to a delay in the DTLS connection handshake between Asterisk and the browser (although this is just a guess after trying to isolate the issue).
> sip.conf
> {code}
> [100]
> nat=force_rport,comedia
> host=dynamic
> type=friend
> secret=secret
> disallow=all
> allow=g722
> icesupport=yes
> transport=wss
> dtlsenable=yes
> dtlsverify=no
> dtlscertfile=/cert.crt
> dtlsprivatekey=/cert.key
> dtlssetup=actpass
> videosupport=no
> encryption=yes
> avpf=yes
> force_avp=yes
> directmedia=no
> canreinvite=no
> context=wrtc
> {code}
> extensions.conf
> {code}
> [wrtc]
> exten => _X.,1,Answer
> same => n,Playback(tt-monkeys&tt-monkeys&tt-monkeys&tt-monkeys&tt-monkeys&tt-monkeys&tt-monkeys&tt-monkeys&tt-monkeys&tt-monkeys&tt-monkeys&tt-monkeys&tt-monkeys&tt-monkeys&tt-monkeys&tt-monkeys&tt-monkeys)
> same => n,Hangup
> {code}
> rtp.conf
> {code}
> [general]
> rtpstart=6000
> rtpend=65535
> icesupport=true
> [ice_host_candidates]
> x.x.x.x => y.y.y.y ; x.x.x.x is the internal IP, y.y.y.y is the external IP
> {code}
> {code}
> <--- SIP read from WS:BROWSER-EXTERNAL-IP:57667 --->
> INVITE sip:123456 at WEBRTC-FQDN SIP/2.0
> Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKEwnJD94ZuargVyxGymxlcyBIreeXDvTf;rport
> From: "100"<sip:100 at WEBRTC-FQDN>;tag=6q5wNcGS9CnWOkMKZ92N
> To: <sip:123456 at WEBRTC-FQDN>
> Contact: "100"<sips:100 at df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=wss>;impi=100;ha1=e18a080ecd528487deae793cb4eaa28d;+g.oma.sip-im;language="en,fr"
> Call-ID: 672fae6f-99fd-de1c-0347-fd15e703483c
> CSeq: 31996 INVITE
> Content-Type: application/sdp
> Content-Length: 2550
> Max-Forwards: 70
> User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
> Organization: Doubango Telecom
> v=0
> o=- 721192063149443000 2 IN IP4 127.0.0.1
> s=Doubango Telecom - chrome
> t=0 0
> a=group:BUNDLE audio
> a=msid-semantic: WMS AggXYiBL3BB5upWZUJP1vuubyWvp2VBdpAFk
> m=audio 52335 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
> c=IN IP4 BROWSER-EXTERNAL-IP
> a=rtcp:60108 IN IP4 BROWSER-EXTERNAL-IP
> a=candidate:2772985396 1 udp 2122265343 BROWSER-INTERNAL-IPv6 64179 typ host generation 0 network-id 2 network-cost 50
> a=candidate:430735571 1 udp 2122194687 BROWSER-INTERNAL-IP 52335 typ host generation 0 network-id 1 network-cost 10
> a=candidate:2772985396 2 udp 2122265342 BROWSER-INTERNAL-IPv6 52336 typ host generation 0 network-id 2 network-cost 50
> a=candidate:430735571 2 udp 2122194686 BROWSER-INTERNAL-IP 60108 typ host generation 0 network-id 1 network-cost 10
> a=candidate:3955989188 1 tcp 1518285567 BROWSER-INTERNAL-IPv6 9 typ host tcptype active generation 0 network-id 2 network-cost 50
> a=candidate:1462729763 1 tcp 1518214911 BROWSER-INTERNAL-IP 9 typ host tcptype active generation 0 network-id 1 network-cost 10
> a=candidate:3955989188 2 tcp 1518285566 BROWSER-INTERNAL-IPv6 9 typ host tcptype active generation 0 network-id 2 network-cost 50
> a=candidate:1462729763 2 tcp 1518214910 BROWSER-INTERNAL-IP 9 typ host tcptype active generation 0 network-id 1 network-cost 10
> a=candidate:2565113447 1 udp 1685987071 BROWSER-EXTERNAL-IP 52335 typ srflx raddr BROWSER-INTERNAL-IP rport 52335 generation 0 network-id 1 network-cost 10
> a=candidate:2565113447 2 udp 1685987070 BROWSER-EXTERNAL-IP 60108 typ srflx raddr BROWSER-INTERNAL-IP rport 60108 generation 0 network-id 1 network-cost 10
> a=ice-ufrag:Cd81
> a=ice-pwd:awZyRbaK+C6JjzrNKY0eNTk8
> a=fingerprint:sha-256 EA:7E:95:E2:50:52:7C:A7:A2:DA:D4:6D:8F:30:BD:5A:C6:A0:56:DC:3E:7B:ED:37:59:94:9B:DD:2E:4D:2B:8B
> a=setup:actpass
> a=mid:audio
> a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
> a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
> a=sendrecv
> a=rtcp-mux
> a=rtpmap:111 opus/48000/2
> a=rtcp-fb:111 transport-cc
> a=fmtp:111 minptime=10;useinbandfec=1
> a=rtpmap:103 ISAC/16000
> a=rtpmap:104 ISAC/32000
> a=rtpmap:9 G722/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:106 CN/32000
> a=rtpmap:105 CN/16000
> a=rtpmap:13 CN/8000
> a=rtpmap:126 telephone-event/8000
> a=ssrc:3291094471 cname:43c4bgpLbedB78m/
> a=ssrc:3291094471 msid:AggXYiBL3BB5upWZUJP1vuubyWvp2VBdpAFk ce18a2c5-5612-4176-bee2-b9590533bb50
> a=ssrc:3291094471 mslabel:AggXYiBL3BB5upWZUJP1vuubyWvp2VBdpAFk
> a=ssrc:3291094471 label:ce18a2c5-5612-4176-bee2-b9590533bb50
> <------------->
> --- (12 headers 44 lines) ---
> Using INVITE request as basis request - 672fae6f-99fd-de1c-0347-fd15e703483c
> Found peer '100' for '100' from BROWSER-EXTERNAL-IP:57667
> <--- Reliably Transmitting (NAT) to BROWSER-EXTERNAL-IP:57667 --->
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKEwnJD94ZuargVyxGymxlcyBIreeXDvTf;received=BROWSER-EXTERNAL-IP;rport=57667
> From: "100"<sip:100 at WEBRTC-FQDN>;tag=6q5wNcGS9CnWOkMKZ92N
> To: <sip:123456 at WEBRTC-FQDN>;tag=as0fe40e33
> Call-ID: 672fae6f-99fd-de1c-0347-fd15e703483c
> CSeq: 31996 INVITE
> Server: Asterisk PBX 14.1.1
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="23bf6e73"
> Content-Length: 0
> <------------>
> Scheduling destruction of SIP dialog '672fae6f-99fd-de1c-0347-fd15e703483c' in 32000 ms (Method: INVITE)
> <--- SIP read from WS:BROWSER-EXTERNAL-IP:57667 --->
> ACK sip:123456 at WEBRTC-FQDN SIP/2.0
> Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKEwnJD94ZuargVyxGymxlcyBIreeXDvTf;rport
> From: "100"<sip:100 at WEBRTC-FQDN>;tag=6q5wNcGS9CnWOkMKZ92N
> To: <sip:123456 at WEBRTC-FQDN>;tag=as0fe40e33
> Call-ID: 672fae6f-99fd-de1c-0347-fd15e703483c
> CSeq: 31996 ACK
> Content-Length: 0
> Max-Forwards: 70
> <------------->
> --- (8 headers 0 lines) ---
> <--- SIP read from WS:BROWSER-EXTERNAL-IP:57667 --->
> INVITE sip:123456 at WEBRTC-FQDN SIP/2.0
> Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKXO1rg8w6ZgHQTmzbguUbIwgTaIV6zLrV;rport
> From: "100"<sip:100 at WEBRTC-FQDN>;tag=6q5wNcGS9CnWOkMKZ92N
> To: <sip:123456 at WEBRTC-FQDN>
> Contact: "100"<sips:100 at df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=wss>;impi=100;ha1=e18a080ecd528487deae793cb4eaa28d;+g.oma.sip-im;language="en,fr"
> Call-ID: 672fae6f-99fd-de1c-0347-fd15e703483c
> CSeq: 31997 INVITE
> Content-Type: application/sdp
> Content-Length: 2550
> Max-Forwards: 70
> Authorization: Digest username="100",realm="asterisk",nonce="23bf6e73",uri="sip:123456 at WEBRTC-FQDN",response="63b58112d226aa95421aa24f12048d1c",algorithm=MD5
> User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
> Organization: Doubango Telecom
> v=0
> o=- 721192063149443000 2 IN IP4 127.0.0.1
> s=Doubango Telecom - chrome
> t=0 0
> a=group:BUNDLE audio
> a=msid-semantic: WMS AggXYiBL3BB5upWZUJP1vuubyWvp2VBdpAFk
> m=audio 52335 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
> c=IN IP4 BROWSER-EXTERNAL-IP
> a=rtcp:60108 IN IP4 BROWSER-EXTERNAL-IP
> a=candidate:2772985396 1 udp 2122265343 BROWSER-INTERNAL-IPv6 64179 typ host generation 0 network-id 2 network-cost 50
> a=candidate:430735571 1 udp 2122194687 BROWSER-INTERNAL-IP 52335 typ host generation 0 network-id 1 network-cost 10
> a=candidate:2772985396 2 udp 2122265342 BROWSER-INTERNAL-IPv6 52336 typ host generation 0 network-id 2 network-cost 50
> a=candidate:430735571 2 udp 2122194686 BROWSER-INTERNAL-IP 60108 typ host generation 0 network-id 1 network-cost 10
> a=candidate:3955989188 1 tcp 1518285567 BROWSER-INTERNAL-IPv6 9 typ host tcptype active generation 0 network-id 2 network-cost 50
> a=candidate:1462729763 1 tcp 1518214911 BROWSER-INTERNAL-IP 9 typ host tcptype active generation 0 network-id 1 network-cost 10
> a=candidate:3955989188 2 tcp 1518285566 BROWSER-INTERNAL-IPv6 9 typ host tcptype active generation 0 network-id 2 network-cost 50
> a=candidate:1462729763 2 tcp 1518214910 BROWSER-INTERNAL-IP 9 typ host tcptype active generation 0 network-id 1 network-cost 10
> a=candidate:2565113447 1 udp 1685987071 BROWSER-EXTERNAL-IP 52335 typ srflx raddr BROWSER-INTERNAL-IP rport 52335 generation 0 network-id 1 network-cost 10
> a=candidate:2565113447 2 udp 1685987070 BROWSER-EXTERNAL-IP 60108 typ srflx raddr BROWSER-INTERNAL-IP rport 60108 generation 0 network-id 1 network-cost 10
> a=ice-ufrag:Cd81
> a=ice-pwd:awZyRbaK+C6JjzrNKY0eNTk8
> a=fingerprint:sha-256 EA:7E:95:E2:50:52:7C:A7:A2:DA:D4:6D:8F:30:BD:5A:C6:A0:56:DC:3E:7B:ED:37:59:94:9B:DD:2E:4D:2B:8B
> a=setup:actpass
> a=mid:audio
> a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
> a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
> a=sendrecv
> a=rtcp-mux
> a=rtpmap:111 opus/48000/2
> a=rtcp-fb:111 transport-cc
> a=fmtp:111 minptime=10;useinbandfec=1
> a=rtpmap:103 ISAC/16000
> a=rtpmap:104 ISAC/32000
> a=rtpmap:9 G722/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:106 CN/32000
> a=rtpmap:105 CN/16000
> a=rtpmap:13 CN/8000
> a=rtpmap:126 telephone-event/8000
> a=ssrc:3291094471 cname:43c4bgpLbedB78m/
> a=ssrc:3291094471 msid:AggXYiBL3BB5upWZUJP1vuubyWvp2VBdpAFk ce18a2c5-5612-4176-bee2-b9590533bb50
> a=ssrc:3291094471 mslabel:AggXYiBL3BB5upWZUJP1vuubyWvp2VBdpAFk
> a=ssrc:3291094471 label:ce18a2c5-5612-4176-bee2-b9590533bb50
> <------------->
> --- (13 headers 44 lines) ---
> Using INVITE request as basis request - 672fae6f-99fd-de1c-0347-fd15e703483c
> Found peer '100' for '100' from BROWSER-EXTERNAL-IP:57667
> == DTLS ECDH initialized (automatic), faster PFS enabled
> == Using SIP RTP CoS mark 5
> Found RTP audio format 111
> Found RTP audio format 103
> Found RTP audio format 104
> Found RTP audio format 9
> Found RTP audio format 0
> Found RTP audio format 8
> Found RTP audio format 106
> Found RTP audio format 105
> Found RTP audio format 13
> Found RTP audio format 126
> Found audio description format opus for ID 111
> Found unknown media description format ISAC for ID 103
> Found unknown media description format ISAC for ID 104
> Found audio description format G722 for ID 9
> Found audio description format PCMU for ID 0
> Found audio description format PCMA for ID 8
> Found unknown media description format CN for ID 106
> Found unknown media description format CN for ID 105
> Found audio description format CN for ID 13
> Found audio description format telephone-event for ID 126
> Capabilities: us - (g722), peer - audio=(ulaw|alaw|g722|opus)/video=(nothing)/text=(nothing), combined - (g722)
> Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)
> Peer audio RTP is at port BROWSER-EXTERNAL-IP:52335
> Looking for 123456 in wrtc (domain WEBRTC-FQDN)
> sip_route_dump: route/path hop: <sips:100 at df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=wss>
> <--- Transmitting (NAT) to BROWSER-EXTERNAL-IP:57667 --->
> SIP/2.0 100 Trying
> Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKXO1rg8w6ZgHQTmzbguUbIwgTaIV6zLrV;received=BROWSER-EXTERNAL-IP;rport=57667
> From: "100"<sip:100 at WEBRTC-FQDN>;tag=6q5wNcGS9CnWOkMKZ92N
> To: <sip:123456 at WEBRTC-FQDN>
> Call-ID: 672fae6f-99fd-de1c-0347-fd15e703483c
> CSeq: 31997 INVITE
> Server: Asterisk PBX 14.1.1
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Contact: <sip:123456 at ASTERISK-INTERNAL-IP:5060;transport=WS>
> Content-Length: 0
> <------------>
> -- Executing [123456 at wrtc:1] Answer("SIP/100-00000001", "") in new stack
> Audio is at 52848
> Adding codec g722 to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
> <--- Reliably Transmitting (NAT) to BROWSER-EXTERNAL-IP:57667 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKXO1rg8w6ZgHQTmzbguUbIwgTaIV6zLrV;received=BROWSER-EXTERNAL-IP;rport=57667
> From: "100"<sip:100 at WEBRTC-FQDN>;tag=6q5wNcGS9CnWOkMKZ92N
> To: <sip:123456 at WEBRTC-FQDN>;tag=as2984a23e
> Call-ID: 672fae6f-99fd-de1c-0347-fd15e703483c
> CSeq: 31997 INVITE
> Server: Asterisk PBX 14.1.1
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Contact: <sip:123456 at ASTERISK-INTERNAL-IP:5060;transport=WS>
> Content-Type: application/sdp
> Content-Length: 624
> v=0
> o=root 656480106 656480106 IN IP4 ASTERISK-INTERNAL-IP
> s=Asterisk PBX 14.1.1
> c=IN IP4 ASTERISK-INTERNAL-IP
> t=0 0
> m=audio 52848 RTP/SAVPF 9 126
> a=rtpmap:9 G722/8000
> a=rtpmap:126 telephone-event/8000
> a=fmtp:126 0-16
> a=maxptime:150
> a=ice-ufrag:6db2cf0f09874f0d027752705f609348
> a=ice-pwd:2f24f51c3e0a1dbf5edfd1370fc02910
> a=candidate:H34ca6dca 1 UDP 2130706431 ASTERISK-EXTERNAL-IP 52848 typ host
> a=candidate:H34ca6dca 2 UDP 2130706430 ASTERISK-EXTERNAL-IP 52849 typ host
> a=connection:new
> a=setup:active
> a=fingerprint:SHA-256 35:BF:17:2A:09:FB:6E:26:75:DE:CE:A1:DE:02:AC:E8:11:55:4F:BA:39:2B:8B:45:51:D6:9E:6F:2E:80:83:4B
> a=sendrecv
> <------------>
> <--- SIP read from WS:BROWSER-EXTERNAL-IP:57667 --->
> ACK sip:123456 at ASTERISK-INTERNAL-IP:5060;transport=WS SIP/2.0
> Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKT3Di3kyStbskrP6Cq0SA;rport
> From: "100"<sip:100 at WEBRTC-FQDN>;tag=6q5wNcGS9CnWOkMKZ92N
> To: <sip:123456 at WEBRTC-FQDN>;tag=as2984a23e
> Contact: "100"<sips:100 at df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=wss>;+g.oma.sip-im;language="en,fr"
> Call-ID: 672fae6f-99fd-de1c-0347-fd15e703483c
> CSeq: 31997 ACK
> Content-Length: 0
> Max-Forwards: 70
> Authorization: Digest username="100",realm="asterisk",nonce="23bf6e73",uri="sip:123456 at ASTERISK-INTERNAL-IP:5060;transport=WS",response="9a503410d8ae4acf4de2b073a5062d51",algorithm=MD5
> User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
> Organization: Doubango Telecom
> <------------->
> --- (12 headers 0 lines) ---
> -- Executing [123456 at wrtc:2] Playback("SIP/100-00000001", "tt-monkeys&tt-monkeys&tt-monkeys&tt-monkeys&tt-monkeys&tt-monkeys&tt-monkeys&tt-monkeys&tt-monkeys&tt-monkeys&tt-monkeys&tt-monkeys&tt-monkeys&tt-monkeys&tt-monkeys&tt-monkeys&tt-monkeys") in new stack
> Sent RTP packet to BROWSER-EXTERNAL-IP:52335 (via ICE) (type 09, seq 004365, ts 000160, len 000160)
> -- <SIP/100-00000001> Playing 'tt-monkeys.gsm' (language 'en')
> Sent RTP packet to BROWSER-EXTERNAL-IP:52335 (via ICE) (type 09, seq 004366, ts 000320, len 000160)
> Sent RTP packet to BROWSER-EXTERNAL-IP:52335 (via ICE) (type 09, seq 004367, ts 000480, len 000160)
> Sent RTP packet to BROWSER-EXTERNAL-IP:52335 (via ICE) (type 09, seq 004368, ts 000640, len 000160)
> Sent RTP packet to BROWSER-EXTERNAL-IP:52335 (via ICE) (type 09, seq 004369, ts 000800, len 000160)
> Sent RTP packet to BROWSER-EXTERNAL-IP:52335 (via ICE) (type 09, seq 004370, ts 000960, len 000160)
> Sent RTP packet to BROWSER-EXTERNAL-IP:52335 (via ICE) (type 09, seq 004371, ts 001120, len 000160)
> Sent RTP packet to BROWSER-EXTERNAL-IP:52335 (via ICE) (type 09, seq 004372, ts 001280, len 000160)
> Sent RTP packet to BROWSER-EXTERNAL-IP:52335 (via ICE) (type 09, seq 004373, ts 001440, len 000160)
> Sent RTP packet to BROWSER-EXTERNAL-IP:52335 (via ICE) (type 09, seq 004374, ts 001600, len 000160)
> {code}
> FireFox shows this in its logs:
> {code}
> 276467712[7f8929a4aec0]: [|WebrtcAudioSessionConduit] AudioConduit.cpp:612: GetAudioFrame
> 276467712[7f8929a4aec0]: [|WebrtcAudioSessionConduit] AudioConduit.cpp:716: GetAudioFrame GetAudioFrame:Got samples: length 320
> {code}
> But no audio is played until some (random) number of seconds pass, and this logs shows up:
> {code}
> 261443584[7f89265334a0]: Flow[568ce410feebc53d:0,rtp(none)]; Layer[dtls]: PacketReceived(2001)
> 261443584[7f89265334a0]: Flow[568ce410feebc53d:0,rtp(none)]; Layer[dtls]: Checking digest, algorithm=sha-256
> 261443584[7f89265334a0]: Flow[568ce410feebc53d:0,rtp(none)]; Layer[ice]: SendPacket(75) succeeded
> 261443584[7f89265334a0]: Flow[568ce410feebc53d:0,rtp(none)]; Layer[dtls]: ****** SSL handshake completed ******
> 261443584[7f89265334a0]: Flow[568ce410feebc53d:0,rtp(none)]; Layer[dtls]: ALPN not negotiated, selecting default
> 261443584[7f89265334a0]: /builds/slave/m-rel-m64-00000000000000000000/build/src/media/mtransport/transportlayerdtls.cpp:865: Flow[568ce410feebc53d:0,rtp(none)];
> {code}
> Without changing any browser or network settings whatsoever either in the Asterisk Box or the browser's box, sometimes it just works. But most of the time the delay is present.
> With FreeSWITCH (cec0cb39830546a3a1c1df7ad7a05b05f14b8975 - Fri Oct 28 15:38:25 2016 -0500) works perfectly, every single time.
> Any help is greatly appreciated!
> Thank you.
> Best regards,
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