Invalid DTLS-SRTP configuration on RTP instance

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Invalid DTLS-SRTP configuration on RTP instance

Postby edslopes » Mon Jan 04, 2016 12:58 pm

Fellows,

I'm facing this issue on an WebRTC implementation on Asterisk 11.13. Please, does anybody know how to fix it?

[2016-01-03 19:36:50] ERROR[5395][C-00000002]: chan_sip.c:5707 dialog_initialize_dtls_srtp: Attempted to set an invalid DTLS-SRTP configuration on RTP instance '0x7f2eb0032ed8'


Regards,

Edson
edslopes
Newsterisk
 
Posts: 3
Joined: Sun Jun 29, 2014 3:00 pm

Re: Invalid DTLS-SRTP configuration on RTP instance

Postby jcolp » Mon Jan 04, 2016 2:16 pm

I'd suggest posting a complete console output of an attempt as well as your sip.conf. Something about your configuration is not correct, or is incomplete.
Joshua Colp
Digium, Inc. | Senior Software Developer
jcolp
Oldsterisk
 
Posts: 248
Joined: Tue May 19, 2015 6:59 am

Re: Invalid DTLS-SRTP configuration on RTP instance

Postby edslopes » Mon Jan 04, 2016 3:46 pm

Thanks.

As you requested bellow, following files regarding this configuration:

sip.conf:

[general]
tos_sip=cs3
tos_audio=ef
tos_video=af41
rtcachefriends=yes
callcounter=yes
alwaysauthreject=yes
maxexpiry=3600
minexpiry=60
defaultexpiry=120
qualifyfreq=60
qualifygap=100
registertimeout=20
registerattempts=0
rtptimeout=60
rtpholdtimeout=300
videosupport=yes
maxcallbitrate=384
faxdetect=yes
t38pt_udptl=yes
directmedia=no
notifyringing=yes
notifyhold=yes
dtmfmode=auto
relaxdtmf=yes
trustrpid=yes
sendrpid=no
useragent=Elastix 3.0
vmexten=*97
language=en
disallow=all
allow=ulaw
allow=alaw
allow=gsm
g726nonstandard=en
accept_outofcall_message=yes
auth_message_requests=yes
bindaddr=127.0.0.1
bindport=5080
outboundproxy=127.0.0.1
outboundproxyport=5060

sip_custom.conf

[1000]
context=xxxxxxxxxxxx
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
qualify=yes
qualifyfreq=600
transport=udp,ws.wss
encryption=yes
callcounter=yes
force_avp=yes
avpf=yes
icesupport=yes
directmedia=no
dtlsenable=yes
;dtlsbindaddr=0.0.0.0
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlscafile=/etc/asterisk/keys/ca.crt
dtlsprivatekey=/etc/asterisk/keys/asterisk.pem
;dtlscipher=ALL
;dtlsclientmethod=tlsv1
dtlsetup=actpass
dtlsverify=no
;realm= XXXXXXXXXXXX

There it is a sip and asterisk debug together regarding a webrtc call placed on sipml5.org client:


<--- SIP read from WS:<My WebRTC Public IP> :21301 --->
INVITE sip:1001@<My PBX Public IP> SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKVMyq7Xcmqcyeolaser3wmGZwgVB55EDE;rport
From: "1000"<sip:1000@<My PBX Public IP>>;tag=PMBRttNQbAWwBAtOM8PP
To: <sip:1001@<My PBX Public IP>>
Contact: "1000"<sip:1000@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr"
Call-ID: 013c003a-3153-2710-1b71-88612bd34175
CSeq: 5840 INVITE
Content-Type: application/sdp
Content-Length: 2501
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18
Organization: Doubango Telecom

v=0
o=- 7272755989382906000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS A6P0yUYZ5D6Hh8Vj7pivKPMVm2BhgRxPco67
m=audio 21311 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
c=IN IP4 <My WebRTC Public IP>
a=rtcp:21312 IN IP4 <My WebRTC Public IP>
a=candidate:3578476384 1 udp 2122260223 10.0.2.2 58927 typ host generation 0
a=candidate:663000344 1 udp 2122194687 192.168.1.203 58928 typ host generation 0
a=candidate:1221703924 1 udp 2122129151 192.168.0.105 58929 typ host generation 0
a=candidate:3578476384 2 udp 2122260222 10.0.2.2 58930 typ host generation 0
a=candidate:663000344 2 udp 2122194686 192.168.1.203 58931 typ host generation 0
a=candidate:1221703924 2 udp 2122129150 192.168.0.105 58932 typ host generation 0
a=candidate:3382296128 1 udp 1685921535 <My WebRTC Public IP> 21311 typ srflx raddr 192.168.0.105 rport 58929 generation 0
a=candidate:3382296128 2 udp 1685921534 <My WebRTC Public IP> 21312 typ srflx raddr 192.168.0.105 rport 58932 generation 0
a=candidate:2613627792 1 tcp 1518280447 10.0.2.2 0 typ host tcptype active generation 0
a=candidate:1762093544 1 tcp 1518214911 192.168.1.203 0 typ host tcptype active generation 0
a=candidate:106054660 1 tcp 1518149375 192.168.0.105 0 typ host tcptype active generation 0
a=candidate:2613627792 2 tcp 1518280446 10.0.2.2 0 typ host tcptype active generation 0
a=candidate:1762093544 2 tcp 1518214910 192.168.1.203 0 typ host tcptype active generation 0
a=candidate:106054660 2 tcp 1518149374 192.168.0.105 0 typ host tcptype active generation 0
a=ice-ufrag:qef+hJ8LKbmxCNyQ
a=ice-pwd:AVtyOi5HMOkD9XJXr+gJrWla
a=fingerprint:sha-256 B5:5E:9A:C1:6F:2D:3D:74:D2:15:17:B8:F4:B5:A5:D1:44:6A:0E:9C:BC:51:19:8E:90:07:58:9A:FC:49:64:C5
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-h ... -send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
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=maxptime:60
a=ssrc:1162162161 cname:egllZJyD4/lxbs4l
a=ssrc:1162162161 msid:A6P0yUYZ5D6Hh8Vj7pivKPMVm2BhgRxPco67 762c913f-b823-4a54-a560-2507e1a38a7d
a=ssrc:1162162161 mslabel:A6P0yUYZ5D6Hh8Vj7pivKPMVm2BhgRxPco67
a=ssrc:1162162161 label:762c913f-b823-4a54-a560-2507e1a38a7d
<------------->
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Header 0 [ 35]: INVITE sip:1001@<My PBX Public IP> SIP/2.0
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Header 1 [ 89]: Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKVMyq7Xcmqcyeolaser3wmGZwgVB55EDE;rport
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Header 2 [ 59]: From: "1000"<sip:1000@<My PBX Public IP>>;tag=PMBRttNQbAWwBAtOM8PP
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Header 3 [ 26]: To: <sip:1001@<My PBX Public IP>>
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Header 4 [122]: Contact: "1000"<sip:1000@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr"
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Header 5 [ 45]: Call-ID: 013c003a-3153-2710-1b71-88612bd34175
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Header 6 [ 17]: CSeq: 5840 INVITE
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Header 7 [ 29]: Content-Type: application/sdp
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Header 8 [ 20]: Content-Length: 2501
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Header 9 [ 16]: Max-Forwards: 70
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Header 10 [ 49]: User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Header 11 [ 30]: Organization: Doubango Telecom
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Header 12 [ 0]:
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 0 [ 3]: v=0
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 1 [ 42]: o=- 7272755989382906000 2 IN IP4 127.0.0.1
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 2 [ 27]: s=Doubango Telecom - chrome
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 3 [ 5]: t=0 0
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 4 [ 20]: a=group:BUNDLE audio
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 5 [ 57]: a=msid-semantic: WMS A6P0yUYZ5D6Hh8Vj7pivKPMVm2BhgRxPco67
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 6 [ 64]: m=audio 21311 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 7 [ 24]: c=IN IP4 <My WebRTC Public IP>
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 8 [ 35]: a=rtcp:21312 IN IP4 <My WebRTC Public IP>
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 9 [ 76]: a=candidate:3578476384 1 udp 2122260223 10.0.2.2 58927 typ host generation 0
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 10 [ 80]: a=candidate:663000344 1 udp 2122194687 192.168.1.203 58928 typ host generation 0
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 11 [ 81]: a=candidate:1221703924 1 udp 2122129151 192.168.0.105 58929 typ host generation 0
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 12 [ 76]: a=candidate:3578476384 2 udp 2122260222 10.0.2.2 58930 typ host generation 0
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 13 [ 80]: a=candidate:663000344 2 udp 2122194686 192.168.1.203 58931 typ host generation 0
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 14 [ 81]: a=candidate:1221703924 2 udp 2122129150 192.168.0.105 58932 typ host generation 0
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 15 [116]: a=candidate:3382296128 1 udp 1685921535 <My WebRTC Public IP> 21311 typ srflx raddr 192.168.0.105 rport 58929 generation 0
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 16 [116]: a=candidate:3382296128 2 udp 1685921534 <My WebRTC Public IP> 21312 typ srflx raddr 192.168.0.105 rport 58932 generation 0
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 17 [ 87]: a=candidate:2613627792 1 tcp 1518280447 10.0.2.2 0 typ host tcptype active generation 0
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 18 [ 92]: a=candidate:1762093544 1 tcp 1518214911 192.168.1.203 0 typ host tcptype active generation 0
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 19 [ 91]: a=candidate:106054660 1 tcp 1518149375 192.168.0.105 0 typ host tcptype active generation 0
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 20 [ 87]: a=candidate:2613627792 2 tcp 1518280446 10.0.2.2 0 typ host tcptype active generation 0
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 21 [ 92]: a=candidate:1762093544 2 tcp 1518214910 192.168.1.203 0 typ host tcptype active generation 0
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 22 [ 91]: a=candidate:106054660 2 tcp 1518149374 192.168.0.105 0 typ host tcptype active generation 0
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 23 [ 28]: a=ice-ufrag:qef+hJ8LKbmxCNyQ
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 24 [ 34]: a=ice-pwd:AVtyOi5HMOkD9XJXr+gJrWla
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 25 [117]: a=fingerprint:sha-256 B5:5E:9A:C1:6F:2D:3D:74:D2:15:17:B8:F4:B5:A5:D1:44:6A:0E:9C:BC:51:19:8E:90:07:58:9A:FC:49:64:C5
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 26 [ 15]: a=setup:actpass
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 27 [ 11]: a=mid:audio
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 28 [ 54]: a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 29 [ 69]: a=extmap:3 http://www.webrtc.org/experiments/rtp-h ... -send-time
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 30 [ 10]: a=sendrecv
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 31 [ 10]: a=rtcp-mux
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 32 [ 25]: a=rtpmap:111 opus/48000/2
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 33 [ 38]: a=fmtp:111 minptime=10; useinbandfec=1
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 34 [ 23]: a=rtpmap:103 ISAC/16000
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 35 [ 23]: a=rtpmap:104 ISAC/32000
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 36 [ 20]: a=rtpmap:9 G722/8000
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 37 [ 20]: a=rtpmap:0 PCMU/8000
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 38 [ 20]: a=rtpmap:8 PCMA/8000
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 39 [ 21]: a=rtpmap:106 CN/32000
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 40 [ 21]: a=rtpmap:105 CN/16000
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 41 [ 19]: a=rtpmap:13 CN/8000
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 42 [ 33]: a=rtpmap:126 telephone-event/8000
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 43 [ 13]: a=maxptime:60
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 44 [ 40]: a=ssrc:1162162161 cname:egllZJyD4/lxbs4l
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 45 [ 96]: a=ssrc:1162162161 msid:A6P0yUYZ5D6Hh8Vj7pivKPMVm2BhgRxPco67 762c913f-b823-4a54-a560-2507e1a38a7d
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Body 46 [ 62]: a=ssrc:1162162161 mslabel:A6P0yUYZ5D6Hh8Vj7pivKPMVm2BhgRxPco67
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9522 parse_request: Body 47 [ 60]: a=ssrc:1162162161 label:762c913f-b823-4a54-a560-2507e1a38a7d
--- (12 headers 48 lines) ---
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9035 find_call: = Looking for Call ID: 013c003a-3153-2710-1b71-88612bd34175 (Checking From) --From tag PMBRttNQbAWwBAtOM8PP --To-tag
[2016-01-04 21:36:10] DEBUG[15055]: acl.c:979 ast_ouraddrfor: For destination '<My WebRTC Public IP> ', our source address is '<My PBX Public IP>'.
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:3881 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_WS with address 127.0.0.1:5080
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:8638 sip_alloc: Allocating new SIP dialog for 013c003a-3153-2710-1b71-88612bd34175 - INVITE (No RTP)
[2016-01-04 21:36:10] DEBUG[15055][C-00000003]: chan_sip.c:28208 handle_incoming: **** Received INVITE (5) - Command in SIP INVITE
[2016-01-04 21:36:10] DEBUG[15055][C-00000003]: chan_sip.c:25453 handle_request_invite: Initializing initreq for method INVITE - callid 013c003a-3153-2710-1b71-88612bd34175
Using INVITE request as basis request - 013c003a-3153-2710-1b71-88612bd34175
[2016-01-04 21:36:10] DEBUG[15055][C-00000003]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '<My PBX Public IP>' into...
[2016-01-04 21:36:10] DEBUG[15055][C-00000003]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '<My PBX Public IP>' and port ''.
Found peer '1000' for '1000' from <My WebRTC Public IP> :21301
[2016-01-04 21:36:10] DEBUG[15055][C-00000003]: rtp_engine.c:283 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0x2b48908'
[2016-01-04 21:36:10] DEBUG[15055][C-00000003]: res_rtp_asterisk.c:2415 ast_rtp_new: Allocated port 16182 for RTP instance '0x2b48908'
[2016-01-04 21:36:10] DEBUG[15055][C-00000003]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '<My PBX Public IP>' into...
[2016-01-04 21:36:10] DEBUG[15055][C-00000003]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '<My PBX Public IP>' and port ''.
[2016-01-04 21:36:10] DEBUG[15055][C-00000003]: stun.c:417 ast_stun_request: stun_send try 1 failed: Invalid argument
[2016-01-04 21:36:10] DEBUG[15055][C-00000003]: rtp_engine.c:292 ast_rtp_instance_new: RTP instance '0x2b48908' is setup and ready to go
[2016-01-04 21:36:10] ERROR[15055][C-00000003]: chan_sip.c:5707 dialog_initialize_dtls_srtp: Attempted to set an invalid DTLS-SRTP configuration on RTP instance '0x2b48908'
[2016-01-04 21:36:10] NOTICE[15055][C-00000003]: chan_sip.c:25534 handle_request_invite: Failed to authenticate device "1000"<sip:1000@<My PBX Public IP>>;tag=PMBRttNQbAWwBAtOM8PP


<--- Reliably Transmitting (NAT) to <My WebRTC Public IP> :21301 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKVMyq7Xcmqcyeolaser3wmGZwgVB55EDE;received=<My WebRTC Public IP> ;rport=21301
From: "1000"<sip:1000@<My PBX Public IP>>;tag=PMBRttNQbAWwBAtOM8PP
To: <sip:1001@<My PBX Public IP>>;tag=as196f7476
Call-ID: 013c003a-3153-2710-1b71-88612bd34175
CSeq: 5840 INVITE
Server: Elastix 3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
[2016-01-04 21:36:10] DEBUG[15055][C-00000003]: chan_sip.c:3724 __sip_xmit: Trying to put 'SIP/2.0 403' onto WS socket destined for <My WebRTC Public IP> :21301
Scheduling destruction of SIP dialog '013c003a-3153-2710-1b71-88612bd34175' in 6400 ms (Method: INVITE)
[2016-01-04 21:36:10] DEBUG[15055][C-00000003]: res_config_odbc.c:114 custom_prepare: Skip: 0; SQL: SELECT * FROM sip WHERE name = ? AND host = ?
[2016-01-04 21:36:10] DEBUG[15055][C-00000003]: res_config_odbc.c:130 custom_prepare: Parameter 1 ('name') = '1001'
[2016-01-04 21:36:10] DEBUG[15055][C-00000003]: res_config_odbc.c:130 custom_prepare: Parameter 2 ('host') = 'dynamic'
[2016-01-04 21:36:10] DEBUG[15055][C-00000003]: res_odbc.c:1058 odbc_release_obj2: odbc_release_obj2(0x2a35bf8) called (obj->txf = (nil))
[2016-01-04 21:36:10] DEBUG[15055][C-00000003]: res_config_odbc.c:114 custom_prepare: Skip: 0; SQL: SELECT * FROM sip WHERE name = ?
[2016-01-04 21:36:10] DEBUG[15055][C-00000003]: res_config_odbc.c:130 custom_prepare: Parameter 1 ('name') = '1001'
[2016-01-04 21:36:10] DEBUG[15055][C-00000003]: res_odbc.c:1058 odbc_release_obj2: odbc_release_obj2(0x2a35bf8) called (obj->txf = (nil))

<--- SIP read from WS:<My WebRTC Public IP> :21301 --->
ACK sip:1001@<My PBX Public IP> SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKVMyq7Xcmqcyeolaser3wmGZwgVB55EDE;rport
From: "1000"<sip:1000@<My PBX Public IP>>;tag=PMBRttNQbAWwBAtOM8PP
To: <sip:1001@<My PBX Public IP>>;tag=as196f7476
Call-ID: 013c003a-3153-2710-1b71-88612bd34175
CSeq: 5840 ACK
Content-Length: 0
Max-Forwards: 70

<------------->
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Header 0 [ 32]: ACK sip:1001@<My PBX Public IP> SIP/2.0
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Header 1 [ 89]: Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKVMyq7Xcmqcyeolaser3wmGZwgVB55EDE;rport
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Header 2 [ 59]: From: "1000"<sip:1000@<My PBX Public IP>>;tag=PMBRttNQbAWwBAtOM8PP
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Header 3 [ 41]: To: <sip:1001@<My PBX Public IP>>;tag=as196f7476
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Header 4 [ 45]: Call-ID: 013c003a-3153-2710-1b71-88612bd34175
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Header 5 [ 14]: CSeq: 5840 ACK
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Header 6 [ 17]: Content-Length: 0
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9485 parse_request: Header 7 [ 16]: Max-Forwards: 70
--- (8 headers 0 lines) ---
[2016-01-04 21:36:10] DEBUG[15055]: chan_sip.c:9035 find_call: = Looking for Call ID: 013c003a-3153-2710-1b71-88612bd34175 (Checking From) --From tag PMBRttNQbAWwBAtOM8PP --To-tag as196f7476
[2016-01-04 21:36:10] DEBUG[15055][C-00000003]: chan_sip.c:28208 handle_incoming: **** Received ACK (6) - Command in SIP ACK
[2016-01-04 21:36:10] DEBUG[15055][C-00000003]: chan_sip.c:4416 __sip_ack: Stopping retransmission on '013c003a-3153-2710-1b71-88612bd34175' of Response 5840: Match Not Found
[2016-01-04 21:36:10] DEBUG[14655]: chan_sip.c:6679 sip_destroy: Destroying SIP dialog 013c003a-3153-2710-1b71-88612bd34175
Really destroying SIP dialog '013c003a-3153-2710-1b71-88612bd34175' Method: INVITE
[2016-01-04 21:36:10] DEBUG[14655]: rtp_engine.c:226 instance_destructor: Destroyed RTP instance '0x2b48908'
sh72xvsxpk*CLI> exit
Asterisk cleanly ending (0).
Executing last minute cleanups
Asterisk ending (0).
[root@sh72xvsxpk asterisk]#
edslopes
Newsterisk
 
Posts: 3
Joined: Sun Jun 29, 2014 3:00 pm

Re: Invalid DTLS-SRTP configuration on RTP instance

Postby navaismo » Mon Jan 04, 2016 5:16 pm

*shooting in the dark*

Self Signed Certificates?
navaismo
Salt of the Asterisk
 
Posts: 1610
Joined: Mon Dec 07, 2009 1:30 pm
Location: Mexico City, Mexico

Re: Invalid DTLS-SRTP configuration on RTP instance

Postby jcolp » Tue Jan 05, 2016 6:24 am

You REALLY need the latest Asterisk version. It will print out error messages in more of the failure cases in this scenario telling you what is wrong, as well browsers now work differently than they did when 11.13 was released, so they won't work with it. When it comes to WebRTC you need to use the latest version of Asterisk in most cases UNLESS you ensure every web browser is a specific version you've tested against it.
Joshua Colp
Digium, Inc. | Senior Software Developer
jcolp
Oldsterisk
 
Posts: 248
Joined: Tue May 19, 2015 6:59 am

Re: Invalid DTLS-SRTP configuration on RTP instance

Postby navaismo » Tue Jan 05, 2016 8:50 am

Asterisk versions on Elastix must be upgraded to work with lates webrtc changes.
navaismo
Salt of the Asterisk
 
Posts: 1610
Joined: Mon Dec 07, 2009 1:30 pm
Location: Mexico City, Mexico


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