sipjs and asterisk voice call no audio issue - webrtc
I am using sipjs 0.7.5 version and asterisk 13.12.1 to established call between 2 sipjs call through webRTC. Both sipjs client and asterisk server are in local network. I am giving asterisk log of both sip messages and rtp packet bellow:
My problem is from log I am seeing that RTP packet is getting sent to both end via ICE. but in client browser no audio is playing. i.e. there is no audio in browser.
<--- SIP read from WS:192.168.40.48:10380 --->
INVITE sip:1063#192.168.40.45:5060 SIP/2.0
Via: SIP/2.0/WS 192.0.2.208;branch=z9hG4bK4771627
Max-Forwards: 70
To: <sip:1063#192.168.40.45:5060>
From: "1062" <sip:1062#192.168.40.45:5060>;tag=dc2e904hed
Call-ID: iu2sjhg9scpcrir0slju
CSeq: 8911 INVITE
Contact: <sip:haitt3bm#192.0.2.208;transport=ws;ob>
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Content-Type: application/sdp
Supported: outbound
User-Agent: SIP.js/0.7.5
Content-Length: 1220
v=0
o=mozilla...THIS_IS_SDPARTA-49.0.2 2371358821422302787 0 IN IP4 0.0.0.0
s=-
t=0 0
a=sendrecv
a=fingerprint:sha-256 DD:4B:CC:5D:70:15:CE:34:6B:A7:AF:2A:F8:AA:E8:B8:00:C3:28:F6:C8:9C:03:F5:E6:69:04:D6:81:11:3B:85
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 55294 UDP/TLS/RTP/SAVPF 109 9 0 8
c=IN IP4 192.168.35.1
a=candidate:0 1 UDP 2122252543 192.168.35.1 55294 typ host
a=candidate:1 1 UDP 2122187007 192.168.221.1 55295 typ host
a=candidate:2 1 UDP 2122121471 192.168.40.48 55296 typ host
a=candidate:0 2 UDP 2122252542 192.168.35.1 55297 typ host
a=candidate:1 2 UDP 2122187006 192.168.221.1 55298 typ host
a=candidate:2 2 UDP 2122121470 192.168.40.48 55299 typ host
a=sendrecv
a=end-of-candidates
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=fmtp:109 maxplaybackrate=48000;stereo=1
a=ice-pwd:43428b6027fb801be25dae4fdf05981a
a=ice-ufrag:84b4daa9
a=mid:sdparta_0
a=msid:{aaa3da10-5310-4167-b4b7-8328a8e5ba84} {4bee853d-06c4-4bfd-ae69-319a1994c844}
a=rtcp:55297 IN IP4 192.168.35.1
a=rtcp-mux
a=rtpmap:109 opus/48000/2
a=rtpmap:9 G722/8000/1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=setup:actpass
a=ssrc:1072212720 cname:{a978e140-65a4-49e8-bb8b-2273ab1b6026}
<------------->
--- (13 headers 32 lines) ---
Using INVITE request as basis request - iu2sjhg9scpcrir0slju
Found peer '1062' for '1062' from 192.168.40.48:10380
== DTLS ECDH initialized (automatic), faster PFS enabled
== DTLS ECDH initialized (automatic), faster PFS enabled
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
[Nov 2 10:07:30] NOTICE[64407][C-00000008]: chan_sip.c:10315 process_sdp: Received SAVPF profle in audio offer but AVPF is not enabled, enabling: audio 55294 UDP/TLS/RTP/SAVPF 109 9 0 8
Found RTP audio format 109
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found audio description format opus for ID 109
Found audio description format G722 for ID 9
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|alaw|g722|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.35.1:55294
Peer doesn't provide video
Looking for 1063 in default (domain 192.168.40.45)
sip_route_dump: route/path hop: <sip:haitt3bm#192.0.2.208;transport=ws;ob>
<--- Transmitting (no NAT) to 192.168.40.48:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WS 192.0.2.208;branch=z9hG4bK4771627;received=192.168.40.48
From: "1062" <sip:1062#192.168.40.45:5060>;tag=dc2e904hed
To: <sip:1063#192.168.40.45:5060>
Call-ID: iu2sjhg9scpcrir0slju
CSeq: 8911 INVITE
Server: Asterisk PBX 13.12.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:1063#192.168.40.45:5060;transport=WS>
Content-Length: 0
<------------>
-- Executing [1063#default:1] Dial("SIP/1062-00000010", "SIP/1063") in new stack
== DTLS ECDH initialized (automatic), faster PFS enabled
== DTLS ECDH initialized (automatic), faster PFS enabled
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
Audio is at 14284
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.40.48:10383:
INVITE sip:thqqcmh6#192.0.2.218;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.40.45:5060;branch=z9hG4bK3f05c747
Max-Forwards: 70
From: "1062" <sip:1062#192.168.40.45>;tag=as0cfcb1f6
To: <sip:thqqcmh6#192.0.2.218;transport=ws>
Contact: <sip:1062#192.168.40.45:5060;transport=WS>
Call-ID: 30d68fec7f1ab1ad5f413e2031ad3290#192.168.40.45:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.12.1
Date: Wed, 02 Nov 2016 04:07:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 878
v=0
o=root 1773642892 1773642892 IN IP4 192.168.40.45
s=Asterisk PBX 13.12.1
c=IN IP4 192.168.40.45
t=0 0
m=audio 14284 RTP/SAVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=ice-ufrag:72465290354ec1f13cdfbb24592c36f1
a=ice-pwd:2e3f757f3f18a66e4ce88b32509c65d3
a=candidate:Hc0a8282d 1 UDP 2130706431 192.168.40.45 14284 typ host
a=candidate:Sb6a06f32 1 UDP 1694498815 182.160.111.50 14284 typ srflx raddr 192.168.40.45 rport 14284
a=candidate:Hc0a8282d 2 UDP 2130706430 192.168.40.45 14285 typ host
a=candidate:Sb6a06f32 2 UDP 1694498814 182.160.111.50 14285 typ srflx raddr 192.168.40.45 rport 14285
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 4B:8C:EF:A9:7A:59:B0:3A:73:B2:BF:70:4D:EC:94:B4:03:BD:FA:5C:E0:50:F8:CD:5B:52:CC:21:5E:D2:32:D4
a=sendrecv
---
-- Called SIP/1063
<--- SIP read from WS:192.168.40.48:10383 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WS 192.168.40.45:5060;branch=z9hG4bK3f05c747
To: <sip:thqqcmh6#192.0.2.218;transport=ws>
From: "1062" <sip:1062#192.168.40.45>;tag=as0cfcb1f6
Call-ID: 30d68fec7f1ab1ad5f413e2031ad3290#192.168.40.45:5060
CSeq: 102 INVITE
Supported: outbound
User-Agent: SIP.js/0.7.5
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from WS:192.168.40.48:10383 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WS 192.168.40.45:5060;branch=z9hG4bK3f05c747
To: <sip:thqqcmh6#192.0.2.218;transport=ws>;tag=iliacjnojs
From: "1062" <sip:1062#192.168.40.45>;tag=as0cfcb1f6
Call-ID: 30d68fec7f1ab1ad5f413e2031ad3290#192.168.40.45:5060
CSeq: 102 INVITE
Contact: <sip:thqqcmh6#192.0.2.218;transport=ws>
Supported: outbound
User-Agent: SIP.js/0.7.5
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:thqqcmh6#192.0.2.218;transport=ws>
-- SIP/1063-00000011 is ringing
<--- Transmitting (no NAT) to 192.168.40.48:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WS 192.0.2.208;branch=z9hG4bK4771627;received=192.168.40.48
From: "1062" <sip:1062#192.168.40.45:5060>;tag=dc2e904hed
To: <sip:1063#192.168.40.45:5060>;tag=as6443d323
Call-ID: iu2sjhg9scpcrir0slju
CSeq: 8911 INVITE
Server: Asterisk PBX 13.12.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:1063#192.168.40.45:5060;transport=WS>
Content-Length: 0
<------------>
<--- SIP read from WS:192.168.40.48:10383 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS 192.168.40.45:5060;branch=z9hG4bK3f05c747
To: <sip:thqqcmh6#192.0.2.218;transport=ws>;tag=iliacjnojs
From: "1062" <sip:1062#192.168.40.45>;tag=as0cfcb1f6
Call-ID: 30d68fec7f1ab1ad5f413e2031ad3290#192.168.40.45:5060
CSeq: 102 INVITE
Contact: <sip:thqqcmh6#192.0.2.218;transport=ws>
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound
User-Agent: SIP.js/0.7.5
Content-Type: application/sdp
Content-Length: 2050
v=0
o=- 8316228832686210830 2 IN IP4 127.0.0.1
s=-
t=0 0
a=msid-semantic: WMS Lezzrl9JzDRw0rDdgktZDKQiqy9HrSw4vQzI
m=audio 55300 UDP/TLS/RTP/SAVP 0 8 101
c=IN IP4 192.168.35.1
a=rtcp:55303 IN IP4 192.168.35.1
a=candidate:1468545711 1 udp 2122260223 192.168.35.1 55300 typ host generation 0 network-id 3
a=candidate:4198037383 1 udp 2122194687 192.168.221.1 55301 typ host generation 0 network-id 2
a=candidate:235970510 1 udp 2122129151 192.168.40.48 55302 typ host generation 0 network-id 1 network-cost 10
a=candidate:1468545711 2 udp 2122260222 192.168.35.1 55303 typ host generation 0 network-id 3
a=candidate:4198037383 2 udp 2122194686 192.168.221.1 55304 typ host generation 0 network-id 2
a=candidate:235970510 2 udp 2122129150 192.168.40.48 55305 typ host generation 0 network-id 1 network-cost 10
a=candidate:420202079 1 tcp 1518280447 192.168.35.1 9 typ host tcptype active generation 0 network-id 3
a=candidate:3032157047 1 tcp 1518214911 192.168.221.1 9 typ host tcptype active generation 0 network-id 2
a=candidate:1083401022 1 tcp 1518149375 192.168.40.48 9 typ host tcptype active generation 0 network-id 1 network-cost 10
a=candidate:420202079 2 tcp 1518280446 192.168.35.1 9 typ host tcptype active generation 0 network-id 3
a=candidate:3032157047 2 tcp 1518214910 192.168.221.1 9 typ host tcptype active generation 0 network-id 2
a=candidate:1083401022 2 tcp 1518149374 192.168.40.48 9 typ host tcptype active generation 0 network-id 1 network-cost 10
a=ice-ufrag:Y4fJ
a=ice-pwd:v8rrrqGJU+aLsIg34kJbx9KL
a=fingerprint:sha-256 E0:34:1F:A5:DC:CA:56:F1:D1:70:61:DD:7F:F6:7A:34:F0:D7:9D:53:EF:22:0C:AB:09:63:B9:F5:1F:63:E4:E3
a=setup:active
a=mid:audio
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=ssrc:4040961807 cname:zubAX2k8KRNEKsTO
a=ssrc:4040961807 msid:Lezzrl9JzDRw0rDdgktZDKQiqy9HrSw4vQzI b17cfaa5-2826-4bce-9f23-be489c523d03
a=ssrc:4040961807 mslabel:Lezzrl9JzDRw0rDdgktZDKQiqy9HrSw4vQzI
a=ssrc:4040961807 label:b17cfaa5-2826-4bce-9f23-be489c523d03
<------------->
--- (12 headers 33 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.35.1:55300
sip_route_dump: route/path hop: <sip:thqqcmh6#192.0.2.218;transport=ws>
set_destination: Parsing <sip:thqqcmh6#192.0.2.218;transport=ws> for address/port to send to
set_destination: URI is for WebSocket, we can't set destination
Transmitting (no NAT) to 192.0.2.218:5060:
ACK sip:thqqcmh6#192.0.2.218;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.40.45:5060;branch=z9hG4bK52a8c1f1
Max-Forwards: 70
From: "1062" <sip:1062#192.168.40.45>;tag=as0cfcb1f6
To: <sip:thqqcmh6#192.0.2.218;transport=ws>;tag=iliacjnojs
Contact: <sip:1062#192.168.40.45:5060;transport=WS>
Call-ID: 30d68fec7f1ab1ad5f413e2031ad3290#192.168.40.45:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.12.1
Content-Length: 0
---
-- SIP/1063-00000011 answered SIP/1062-00000010
Audio is at 11134
Adding codec ulaw to SDP
Adding codec alaw to SDP
<--- Reliably Transmitting (no NAT) to 192.168.40.48:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS 192.0.2.208;branch=z9hG4bK4771627;received=192.168.40.48
From: "1062" <sip:1062#192.168.40.45:5060>;tag=dc2e904hed
To: <sip:1063#192.168.40.45:5060>;tag=as6443d323
Call-ID: iu2sjhg9scpcrir0slju
CSeq: 8911 INVITE
Server: Asterisk PBX 13.12.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:1063#192.168.40.45:5060;transport=WS>
Content-Type: application/sdp
Content-Length: 797
v=0
o=root 562796326 562796326 IN IP4 192.168.40.45
s=Asterisk PBX 13.12.1
c=IN IP4 192.168.40.45
t=0 0
m=audio 11134 RTP/SAVPF 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=maxptime:150
a=ice-ufrag:58d669ea38ed577755928de05ad017a0
a=ice-pwd:4513e77c0ac114a14e1f9a654764e1e9
a=candidate:Hc0a8282d 1 UDP 2130706431 192.168.40.45 11134 typ host
a=candidate:Sb6a06f32 1 UDP 1694498815 182.160.111.50 11134 typ srflx raddr 192.168.40.45 rport 11134
a=candidate:Hc0a8282d 2 UDP 2130706430 192.168.40.45 11135 typ host
a=candidate:Sb6a06f32 2 UDP 1694498814 182.160.111.50 11135 typ srflx raddr 192.168.40.45 rport 11135
a=connection:new
a=setup:active
a=fingerprint:SHA-256 4B:8C:EF:A9:7A:59:B0:3A:73:B2:BF:70:4D:EC:94:B4:03:BD:FA:5C:E0:50:F8:CD:5B:52:CC:21:5E:D2:32:D4
a=sendrecv
<------------>
-- Channel SIP/1063-00000011 joined 'simple_bridge' basic-bridge <047baea4-91e3-43fa-99ec-5b49d10b595c>
-- Channel SIP/1062-00000010 joined 'simple_bridge' basic-bridge <047baea4-91e3-43fa-99ec-5b49d10b595c>
> 0x7f243407b350 -- Probation passed - setting RTP source address to 192.168.40.48:55302
Got RTP packet from 192.168.40.48:55302 (type 00, seq 012779, ts 909638055, len 000160)
Got RTP packet from 192.168.40.48:55302 (type 00, seq 012780, ts 909638215, len 000160)
Got RTP packet from 192.168.40.48:55302 (type 00, seq 012781, ts 909638375, len 000160)
Got RTP packet from 192.168.40.48:55302 (type 00, seq 012782, ts 909638535, len 000160)
Got RTP packet from 192.168.40.48:55302 (type 00, seq 012783, ts 909638695, len 000160)
Got RTP packet from 192.168.40.48:55302 (type 00, seq 012784, ts 909638855, len 000160)
Got RTP packet from 192.168.40.48:55302 (type 00, seq 012785, ts 909639015, len 000160)
<--- SIP read from WS:192.168.40.48:10380 --->
ACK sip:1063#192.168.40.45:5060;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.0.2.208;branch=z9hG4bK6228050
Max-Forwards: 70
To: <sip:1063#192.168.40.45:5060>;tag=as6443d323
From: "1062" <sip:1062#192.168.40.45:5060>;tag=dc2e904hed
Call-ID: iu2sjhg9scpcrir0slju
CSeq: 8911 ACK
Supported: outbound
User-Agent: SIP.js/0.7.5
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Got RTP packet from 192.168.40.48:55302 (type 00, seq 012786, ts 909639175, len 000160)
Sent RTP packet to 192.168.35.1:55294 (via ICE) (type 00, seq 006802, ts 909639168, len 000160)
Got RTP packet from 192.168.40.48:55302 (type 00, seq 012787, ts 909639335, len 000160)
Sent RTP packet to 192.168.40.48:55296 (via ICE) (type 00, seq 006803, ts 909639328, len 000170)
> 0x7f24340788f0 -- Probation passed - setting RTP source address to 192.168.40.48:55296
Got RTP packet from 192.168.40.48:55296 (type 00, seq 029688, ts 3658919058, len 000160)
Sent RTP packet to 192.168.40.48:55302 (via ICE) (type 00, seq 024034, ts 3658919056, len 000170)
Got RTP packet from 192.168.40.48:55302 (type 00, seq 012788, ts 909639495, len 000160)
Sent RTP packet to 192.168.40.48:55296 (via ICE) (type 00, seq 006804, ts 909639488, len 000170)
Got RTP packet from 192.168.40.48:55296 (type 00, seq 029689, ts 3658919218, len 000160)
Sent RTP packet to 192.168.40.48:55302 (via ICE) (type 00, seq 024035, ts 3658919216, len 000170)
Got RTP packet from 192.168.40.48:55302 (type 00, seq 012789, ts 909639655, len 000160)
Sent RTP packet to 192.168.40.48:55296 (via ICE) (type 00, seq 006805, ts 909639648, len 000170)
Got RTP packet from 192.168.40.48:55296 (type 00, seq 029690, ts 3658919378, len 000160)
Sent RTP packet to 192.168.40.48:55302 (via ICE) (type 00, seq 024036, ts 3658919376, len 000170)
Got RTP packet from 192.168.40.48:55302 (type 00, seq 012790, ts 909639815, len 000160)
Sent RTP packet to 192.168.40.48:55296 (via ICE) (type 00, seq 006806, ts 909639808, len 000170)
Got RTP packet from 192.168.40.48:55296 (type 00, seq 029691, ts 3658919538, len 000160)
Sent RTP packet to 192.168.40.48:55302 (via ICE) (type 00, seq 024037, ts 3658919536, len 000170)
Got RTP packet from 192.168.40.48:55302 (type 00, seq 012791, ts 909639975, len 000160)
Sent RTP packet to 192.168.40.48:55296 (via ICE) (type 00, seq 006807, ts 909639968, len 000170)
Got RTP packet from 192.168.40.48:55296 (type 00, seq 029692, ts 3658919698, len 000160)
Sent RTP packet to 192.168.40.48:55302 (via ICE) (type 00, seq 024038, ts 3658919696, len 000170)
Got RTP packet from 192.168.40.48:55302 (type 00, seq 012792, ts 909640135, len 000160)
Sent RTP packet to 192.168.40.48:55296 (via ICE) (type 00, seq 006808, ts 909640128, len 000170)
Got RTP packet from 192.168.40.48:55296 (type 00, seq 029693, ts 3658919858, len 000160)
Sent RTP packet to 192.168.40.48:55302 (via ICE) (type 00, seq 024039, ts 3658919856, len 000170)
Got RTP packet from 192.168.40.48:55302 (type 00, seq 012793, ts 909640295, len 000160)
Sent RTP packet to 192.168.40.48:55296 (via ICE) (type 00, seq 006809, ts 909640288, len 000170)
Got RTP packet from 192.168.40.48:55296 (type 00, seq 029694, ts 3658920018, len 000160)
Sent RTP packet to 192.168.40.48:55302 (via ICE) (type 00, seq 024040, ts 3658920016, len 000170)
Got RTP packet from 192.168.40.48:55302 (type 00, seq 012794, ts 909640455, len 000160)
Sent RTP packet to 192.168.40.48:55296 (via ICE) (type 00, seq 006810, ts 909640448, len 000170)
Got RTP packet from 192.168.40.48:55296 (type 00, seq 029695, ts 3658920178, len 000160)
Sent RTP packet to 192.168.40.48:55302 (via ICE) (type 00, seq 024041, ts 3658920176, len 000170)
Got RTP packet from 192.168.40.48:55302 (type 00, seq 012795, ts 909640615, len 000160)
Sent RTP packet to 192.168.40.48:55296 (via ICE) (type 00, seq 006811, ts 909640608, len 000170)
Got RTP packet from 192.168.40.48:55296 (type 00, seq 029696, ts 3658920338, len 000160)
Sent RTP packet to 192.168.40.48:55302 (via ICE) (type 00, seq 024042, ts 3658920336, len 000170)
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FreeSwitch + Mode Verto + Webrtc + Android + unable to make call from android
I have made a mode-verto android client, using WebRtc;
Pre-built library: org.webrtc:google-webrtc:1.0.+
libjingle: io.pristine:libjingle:11139#aar
and FreeSwitch but only got success to make uni-directional communication(SIP phone to an android client, voice communication in both devices successfully). But when I try to make a call from an android client to a SIP phone it doesn't work.
Steps:
From Web To mode-verto android Client(Working scenario):
Login Call:
{"method":"login","id":1,"params":{"passwd":"XXX","userVariables":{},"loginParams":{},"login":"1002#XXX.XXX.com","sessid":"0afc3741-97cf-4b59-94a4-57bdcc2f5f17"},"jsonrpc":"2.0"}
Login Call Response:
{"jsonrpc":"2.0","id":1,"result":{"message":"logged in","sessid":"0afc3741-97cf-4b59-94a4-57bdcc2f5f17"}}
{"jsonrpc":"2.0","id":254,"method":"verto.clientReady","params":{"reattached_sessions":[]}}
Call Invitation From Web:
{"jsonrpc":"2.0","id":255,"method":"verto.invite","params":{"callID":"94c79b9e-9aa1-41d6-85be-3f94d71bf4f4","sdp":"v=0\r\no=FreeSWITCH 1633327591 1633327592 IN IP4 192.168.0.1\r\ns=FreeSWITCH\r\nc=IN IP4 192.168.0.1\r\nt=0 0\r\na=msid-semantic: WMS esmOfEYFBQ88Oj5egBp9rex75ApyzBlE\r\nm=audio 16468 RTP/SAVPF 102 9 0 8\r\na=rtpmap:102 opus/48000/2\r\na=fmtp:102 useinbandfec=1; maxaveragebitrate=30000; maxplaybackrate=48000; ptime=20; minptime=10; maxptime=40; stereo=1\r\na=rtpmap:9 G722/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:8 PCMA/8000\r\na=fingerprint:sha-256 30:D5:C6:44:1E:D6:22:FD:54:D1:0E:E8:7C:B4:E7:AD:93:99:E9:CD:48:D0:6D:B6:28:EE:19:45:83:8A:62:B8\r\na=setup:actpass\r\na=rtcp-mux\r\na=rtcp:16468 IN IP4 192.168.0.1\r\na=ssrc:2708468483 cname:UyjZ27vuNtssDnxJ\r\na=ssrc:2708468483 msid:esmOfEYFBQ88Oj5egBp9rex75ApyzBlE a0\r\na=ssrc:2708468483 mslabel:esmOfEYFBQ88Oj5egBp9rex75ApyzBlE\r\na=ssrc:2708468483 label:esmOfEYFBQ88Oj5egBp9rex75ApyzBlEa0\r\na=ice-ufrag:OMJ6Q73MxiaXJmsu\r\na=ice-pwd:KBkDyFYcZgvtuS29vC5czYOX\r\na=candidate:9757738161 1 udp 2130706431 192.168.0.1 16468 typ host generation 0\r\na=candidate:9757738161 2 udp 2130706431 192.168.0.1 16468 typ host generation 0\r\na=silenceSupp:off - - - -\r\na=ptime:20\r\na=sendrecv\r\nm=video 16470 RTP/SAVPF 96 98 100 102 127 125 108 124 123\r\nb=AS:3072\r\na=rtpmap:96 VP8/90000\r\na=rtpmap:98 VP9/90000\r\na=fmtp:98 profile-id=0\r\na=rtpmap:100 VP9/90000\r\na=fmtp:100 profile-id=2\r\na=rtpmap:102 H264/90000\r\na=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f\r\na=rtpmap:127 H264/90000\r\na=fmtp:127 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42001f\r\na=rtpmap:125 H264/90000\r\na=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f\r\na=rtpmap:108 H264/90000\r\na=fmtp:108 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f\r\na=rtpmap:124 H264/90000\r\na=fmtp:124 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=4d001f\r\na=rtpmap:123 H264/90000\r\na=fmtp:123 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=64001f\r\na=sendrecv\r\na=fingerprint:sha-256 30:D5:C6:44:1E:D6:22:FD:54:D1:0E:E8:7C:B4:E7:AD:93:99:E9:CD:48:D0:6D:B6:28:EE:19:45:83:8A:62:B8\r\na=setup:actpass\r\na=rtcp-mux\r\na=rtcp:16470 IN IP4 192.168.0.1\r\na=rtcp-fb:96 ccm fir\r\na=rtcp-fb:96 ccm tmmbr\r\na=rtcp-fb:96 nack\r\na=rtcp-fb:96 nack pli\r\na=rtcp-fb:98 ccm fir\r\na=rtcp-fb:98 ccm tmmbr\r\na=rtcp-fb:98 nack\r\na=rtcp-fb:98 nack pli\r\na=rtcp-fb:100 ccm fir\r\na=rtcp-fb:100 ccm tmmbr\r\na=rtcp-fb:100 nack\r\na=rtcp-fb:100 nack pli\r\na=rtcp-fb:102 ccm fir\r\na=rtcp-fb:102 ccm tmmbr\r\na=rtcp-fb:102 nack\r\na=rtcp-fb:102 nack pli\r\na=rtcp-fb:127 ccm fir\r\na=rtcp-fb:127 ccm tmmbr\r\na=rtcp-fb:127 nack\r\na=rtcp-fb:127 nack pli\r\na=rtcp-fb:125 ccm fir\r\na=rtcp-fb:125 ccm tmmbr\r\na=rtcp-fb:125 nack\r\na=rtcp-fb:125 nack pli\r\na=rtcp-fb:108 ccm fir\r\na=rtcp-fb:108 ccm tmmbr\r\na=rtcp-fb:108 nack\r\na=rtcp-fb:108 nack pli\r\na=rtcp-fb:124 ccm fir\r\na=rtcp-fb:124 ccm tmmbr\r\na=rtcp-fb:124 nack\r\na=rtcp-fb:124 nack pli\r\na=rtcp-fb:123 ccm fir\r\na=rtcp-fb:123 ccm tmmbr\r\na=rtcp-fb:123 nack\r\na=rtcp-fb:123 nack pli\r\na=ssrc:1891818205 cname:UyjZ27vuNtssDnxJ\r\na=ssrc:1891818205 msid:esmOfEYFBQ88Oj5egBp9rex75ApyzBlE v0\r\na=ssrc:1891818205 mslabel:esmOfEYFBQ88Oj5egBp9rex75ApyzBlE\r\na=ssrc:1891818205 label:esmOfEYFBQ88Oj5egBp9rex75ApyzBlEv0\r\na=ice-ufrag:pWTWGFjyLZJAyUKz\r\na=ice-pwd:0XZUOcLk08IHnyICkspRXyfz\r\na=candidate:0896075528 1 udp 2130706431 192.168.0.1 16470 typ host generation 0\r\na=candidate:0896075528 2 udp 2130706430 192.168.0.1 16470 typ host generation 0\r\na=end-of-candidates\r\n","caller_id_name":"Extension 1001","caller_id_number":"1001","callee_id_name":"Outbound Call","callee_id_number":"1002","display_direction":"outbound"}}
Call Response with SDP:
{"method":"verto.answer","id":2,"params":{"dialogParams":{"callID":"94c79b9e-9aa1-41d6-85be-3f94d71bf4f4","remote_caller_id_number":"1002","destination_number":"1002","remote_caller_id_name":"1002"},"session_id":"0afc3741-97cf-4b59-94a4-57bdcc2f5f17","sdp":"v=0\r\no=- 4285064094073427202 2 IN IP4 127.0.0.1\r\ns=-\r\nt=0 0\r\na=msid-semantic: WMS MediaStream\r\nm=audio 63288 RTP\/SAVPF 102 9 0 8\r\nc=IN IP4 192.168.2.1\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=candidate:341193750 1 udp 2122260223 192.168.8.101 49077 typ host generation 0 network-id 3 network-cost 10\r\na=candidate:1510613869 1 udp 2122129151 127.0.0.1 49460 typ host generation 0 network-id 1\r\na=candidate:842163049 1 udp 1686052607 192.168.2.1 63288 typ srflx raddr 192.168.8.101 rport 49077 generation 0 network-id 3 network-cost 10\r\na=ice-ufrag:BfLO\r\na=ice-pwd:RBsUQ\/uNeEiqmPgyvWJBN4vs\r\na=ice-options:trickle renomination\r\na=fingerprint:sha-256 3D:3A:73:40:49:34:6F:E6:F0:17:9F:E6:7F:FE:C1:0D:C0:3D:86:A1:28:39:73:3A:EB:64:67:E4:57:55:EA:C0\r\na=setup:active\r\na=mid:audio\r\na=sendrecv\r\na=rtcp-mux\r\na=rtpmap:102 opus\/48000\/2\r\na=fmtp:102 minptime=10;useinbandfec=1\r\na=rtpmap:9 G722\/8000\r\na=rtpmap:0 PCMU\/8000\r\na=rtpmap:8 PCMA\/8000\r\na=ssrc:2607788540 cname:lwXiQ5hT1hGGiwou\r\na=ssrc:2607788540 msid:MediaStream ARDAMSa0\r\na=ssrc:2607788540 mslabel:MediaStream\r\na=ssrc:2607788540 label:ARDAMSa0\r\nm=video 0 RTP\/SAVPF 96 98 125\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=ice-ufrag:OxRK\r\na=ice-pwd:LCVdU01PHZJFx4\/dv4j3nqdN\r\na=ice-options:trickle renomination\r\na=fingerprint:sha-256 3D:3A:73:40:49:34:6F:E6:F0:17:9F:E6:7F:FE:C1:0D:C0:3D:86:A1:28:39:73:3A:EB:64:67:E4:57:55:EA:C0\r\na=setup:active\r\na=mid:video\r\na=inactive\r\na=rtcp-mux\r\na=rtpmap:96 VP8\/90000\r\na=rtcp-fb:96 ccm fir\r\na=rtcp-fb:96 nack\r\na=rtcp-fb:96 nack pli\r\na=rtpmap:98 VP9\/90000\r\na=rtcp-fb:98 ccm fir\r\na=rtcp-fb:98 nack\r\na=rtcp-fb:98 nack pli\r\na=rtpmap:125 H264\/90000\r\na=rtcp-fb:125 ccm fir\r\na=rtcp-fb:125 nack\r\na=rtcp-fb:125 nack pli\r\na=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f\r\n"},"jsonrpc":"2.0"}
Followings are the messages received after successful call invitation response:
{"jsonrpc":"2.0","id":2,"result":{"sessid":"0afc3741-97cf-4b59-94a4-57bdcc2f5f17"}}
{"jsonrpc":"2.0","id":259,"method":"verto.display","params":{"callID":"94c79b9e-9aa1-41d6-85be-3f94d71bf4f4","display_name":"Extension 1001","display_number":"1001","caller_id_name":"Extension 1001","caller_id_number":"1001","callee_id_name":"Outbound Call","callee_id_number":"1002","display_direction":"outbound"}}
{"jsonrpc":"2.0","id":260,"method":"verto.bye","params":{"callID":"94c79b9e-9aa1-41d6-85be-3f94d71bf4f4","causeCode":16,"cause":"NORMAL_CLEARING"}}
In the above scenario, the followings states are CONNECTED:
onStandardizedIceConnectionChange
onIceGatheringChange
onConnectionChange
onIceConnectionChange
mode-verto to Web Platform/SIP (Scenario in which help is required):
Login Call:
{"method":"login","id":1,"params":{"passwd":"XXX","userVariables":{},"loginParams":{},"login":"1002#XXX.XXXX.com","sessid":"9743f047-ba15-47ad-828b-26c57529fcf3"},"jsonrpc":"2.0"}
Login Call Response:
{"jsonrpc":"2.0","id":1,"result":{"message":"logged in","sessid":"9743f047-ba15-47ad-828b-26c57529fcf3"}}
{"jsonrpc":"2.0","id":261,"method":"verto.clientReady","params":{"reattached_sessions":[]}}
Call Invitation:
{"method":"verto.invite","id":2,"params":{"dialogParams":{"callID":"50533ec8-985c-4857-99bf-bcb1b021e558","caller_id_name":"1002","remote_caller_id_number":"1001","destination_number":"1001","remote_caller_id_name":"Outbound Call","useVideo":false,"useStereo":false,"useMic":"any","login":"1002#XXXX.XXXX.com","useCamera":"any","screenShare":false,"tag":"any","caller_id_number":"1002","useSpeak":"any","videoParams":{"minHeight":"720","minFrameRate":30,"minWidth":"1280"}},"sdp":"v=0\r\no=- 1985358600062838354 2 IN IP4 127.0.0.1\r\ns=-\r\nt=0 0\r\na=group:BUNDLE audio\r\na=msid-semantic: WMS MediaStream\r\nm=audio 60367 UDP\/TLS\/RTP\/SAVPF 111 103 104 9 102 0 8 106 105 13 110 112 113 126\r\nc=IN IP4 37.111.135.112\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=candidate:341193750 1 udp 2122260223 192.168.2.1 49612 typ host generation 0 network-id 3 network-cost 10\r\na=candidate:1510613869 1 udp 2122129151 127.0.0.1 42121 typ host generation 0 network-id 1\r\na=candidate:842163049 1 udp 1686052607 37.111.135.112 60367 typ srflx raddr 192.168.2.1 rport 49612 generation 0 network-id 3 network-cost 10\r\na=ice-ufrag:DRth\r\na=ice-pwd:6ZrLWyqfDk4vqVPGktrUsCOw\r\na=ice-options:trickle renomination\r\na=fingerprint:sha-256 DF:C1:F2:1E:F9:C6:9F:54:55:CD:A0:78:D5:85:23:B0:C7:E5:6A:C9:48:6C:4D:5E:17:5C:00:D4:F5:D0:D6:BF\r\na=setup:actpass\r\na=mid:audio\r\na=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\na=extmap:2 http:\/\/www.webrtc.org\/experiments\/rtp-hdrext\/abs-send-time\r\na=extmap:3 http:\/\/www.ietf.org\/id\/draft-holmer-rmcat-transport-wide-cc-extensions-01\r\na=sendrecv\r\na=rtcp-mux\r\na=rtpmap:111 opus\/48000\/2\r\na=rtcp-fb:111 transport-cc\r\na=fmtp:111 minptime=10;useinbandfec=1\r\na=rtpmap:103 ISAC\/16000\r\na=rtpmap:104 ISAC\/32000\r\na=rtpmap:9 G722\/8000\r\na=rtpmap:102 ILBC\/8000\r\na=rtpmap:0 PCMU\/8000\r\na=rtpmap:8 PCMA\/8000\r\na=rtpmap:106 CN\/32000\r\na=rtpmap:105 CN\/16000\r\na=rtpmap:13 CN\/8000\r\na=rtpmap:110 telephone-event\/48000\r\na=rtpmap:112 telephone-event\/32000\r\na=rtpmap:113 telephone-event\/16000\r\na=rtpmap:126 telephone-event\/8000\r\na=ssrc:859436163 cname:6xspDJIe1I+kt4ue\r\na=ssrc:859436163 msid:MediaStream ARDAMSa0\r\na=ssrc:859436163 mslabel:MediaStream\r\na=ssrc:859436163 label:ARDAMSa0\r\n","sessid":"9743f047-ba15-47ad-828b-26c57529fcf3"},"jsonrpc":"2.0"}
Call Invitation Response:
{"jsonrpc":"2.0","id":2,"result":{"message":"CALL CREATED","callID":"50533ec8-985c-4857-99bf-bcb1b021e558","sessid":"9743f047-ba15-47ad-828b-26c57529fcf3"}}
{"jsonrpc":"2.0","id":263,"method":"verto.media","params":{"callID":"50533ec8-985c-4857-99bf-bcb1b021e558","sdp":"v=0\r\no=FreeSWITCH 1633328922 1633328923 IN IP4 172.1.0.165\r\ns=FreeSWITCH\r\nc=IN IP4 172.1.0.165\r\nt=0 0\r\na=msid-semantic: WMS YfVYnQ98r94pqu4N6K4IJDNht8vLsWrg\r\nm=audio 17076 UDP/TLS/RTP/SAVPF 111 110\r\na=rtpmap:111 opus/48000/2\r\na=fmtp:111 useinbandfec=1; minptime=10\r\na=rtpmap:110 telephone-event/48000\r\na=silenceSupp:off - - - -\r\na=ptime:20\r\na=sendrecv\r\na=fingerprint:sha-256 30:D5:C6:44:1E:D6:22:FD:54:D1:0E:E8:7C:B4:E7:AD:93:99:E9:CD:48:D0:6D:B6:28:EE:19:45:83:8A:62:B8\r\na=setup:active\r\na=rtcp-mux\r\na=rtcp:17076 IN IP4 172.1.0.165\r\na=ice-ufrag:VRYvkCDEAYgSKxZD\r\na=ice-pwd:zTJiNNjGMzQ6KDmvlbRwEk05\r\na=candidate:8820384080 1 udp 2130706431 172.1.0.165 17076 typ host generation 0\r\na=end-of-candidates\r\na=ssrc:2909163670 cname:2c4fT3eb39qjtkXq\r\na=ssrc:2909163670 msid:YfVYnQ98r94pqu4N6K4IJDNht8vLsWrg a0\r\na=ssrc:2909163670 mslabel:YfVYnQ98r94pqu4N6K4IJDNht8vLsWrg\r\na=ssrc:2909163670 label:YfVYnQ98r94pqu4N6K4IJDNht8vLsWrga0\r\n"}}
Call Invitation Ans. confirmation Response:
{"jsonrpc":"2.0","id":264,"method":"verto.answer","params":{"callID":"50533ec8-985c-4857-99bf-bcb1b021e558"}}
After this process:
onIceConnectionChange and onStandardizedIceConnectionChange get stuck in CHECKING state and onConnectionChange in CONNECTING state.
and after few seconds, onIceConnectionChange , onStandardizedIceConnectionChange and onConnectionChange states change to FAILED.
On the android client, it takes 30, 40 seconds to gather complete Ice-Candidates, which is a lot.
On exchange of successful SDP's , even after ice candidates completion, onIceConnectionChange and onStandardizedIceConnectionChange states change to Failed.
Also followed these solutions:
WebRTC + IOS + Freeswitch : Can't hear audio
https://stackoverflow.com/a/35881785/10413749
No audio when webrtc mobile clients connected in different network
But still, I do not get what I am doing wrong.
Is there anything I'll be missing which I should check? Any help from the community would be really helpful for me.
OpenNebula - Bridge VM NIC with Host NIC - take Ip from LAN DCHP
I hope you are doing well, I start using OpenNebula here, I deploy a basic setup one Opennebula fronend in centos 8 another server as OpenNebula Node, I download an image from marketplace it's centos image, Then I create a network Under Network >> Virual Network. Bridge it with ens33 (ens3 is the physical interface of my node) in order to give VM access to LAN, he is my Node net [centos#host1 ~]$ ifconfig ens33: flags=4163<UP,BROADCAST,RUNNING,MULTICAST> mtu 1500 inet 192.168.0.60 netmask 255.255.255.0 broadcast 192.168.0.255 ether 00:0c:29:68:26:2b txqueuelen 1000 (Ethernet) RX packets 679155 bytes 994474147 (948.4 MiB) RX errors 0 dropped 0 overruns 0 frame 0 TX packets 41914 bytes 3220552 (3.0 MiB) TX errors 0 dropped 0 overruns 0 carrier 0 collisions 0 lo: flags=73<UP,LOOPBACK,RUNNING> mtu 65536 inet 127.0.0.1 netmask 255.0.0.0 inet6 ::1 prefixlen 128 scopeid 0x10<host> loop txqueuelen 1000 (Local Loopback) RX packets 6 bytes 672 (672.0 B) RX errors 0 dropped 0 overruns 0 frame 0 TX packets 6 bytes 672 (672.0 B) TX errors 0 dropped 0 overruns 0 carrier 0 collisions 0 virbr0: flags=4099<UP,BROADCAST,MULTICAST> mtu 1500 inet 192.168.122.1 netmask 255.255.255.0 broadcast 192.168.122.255 ether 52:54:00:89:84:b1 txqueuelen 1000 (Ethernet) RX packets 0 bytes 0 (0.0 B) RX errors 0 dropped 0 overruns 0 frame 0 TX packets 0 bytes 0 (0.0 B) TX errors 0 dropped 0 overruns 0 carrier 0 collisions 0 once I create a VM and attach it to the bridge network I create already, i get status Failed with the bellow log : Sat May 1 03:50:25 2021 [Z0][VM][I]: New state is ACTIVE Sat May 1 03:50:25 2021 [Z0][VM][I]: New LCM state is PROLOG Sat May 1 03:50:38 2021 [Z0][VM][I]: New LCM state is BOOT Sat May 1 03:50:38 2021 [Z0][VMM][I]: Generating deployment file: /var/lib/one/vms/14/deployment.0 Sat May 1 03:50:39 2021 [Z0][VMM][I]: Successfully execute transfer manager driver operation: tm_context. Sat May 1 03:50:40 2021 [Z0][VMM][I]: Command execution fail: cat << EOT | /var/tmp/one/vnm/bridge/pre Sat May 1 03:50:40 2021 [Z0][VMM][E]: pre: Command "sudo ip link add name ens33 type bridge " failed. Sat May 1 03:50:40 2021 [Z0][VMM][E]: pre: RTNETLINK answers: File exists Sat May 1 03:50:40 2021 [Z0][VMM][E]: RTNETLINK answers: File exists Sat May 1 03:50:40 2021 [Z0][VMM][E]: Sat May 1 03:50:40 2021 [Z0][VMM][I]: ExitCode: 2 Sat May 1 03:50:40 2021 [Z0][VMM][I]: Failed to execute network driver operation: pre. Sat May 1 03:50:40 2021 [Z0][VMM][E]: Error deploying virtual machine: bridge: RTNETLINK answers: File exists Sat May 1 03:50:40 2021 [Z0][VM][I]: New LCM state is BOOT_FAILURE can anyone please explain to me what's wrong here, Im familiar with vsphere esxi/vcenter, I want just to create a VMNetwork and attach it to the node physical NIC then attach the VM to this VMNetwork in order to give it LAN access, on VMware side it's easy simple but with OpenNebula Im not sure how it's work Thank you
The problem here is that you are using a physical interface instead of using a bridge. If you would like to use bridge networking, you need to create a bridge or let OpenNebula create it for you. Let me know if this answers your issue, if not, feel free to submit your query on OpenNebula Forum - https://forum.opennebula.io/. :)
Error 488 Not acceptable here
Hello I know there are already lot of topics about this error (or perhaps I don't have the same problem) but none of them answered my question, I am in a local network with blink on my PC and my asterisk server is on an external server hosted by ovh (so there is nat to do). I control the server via encrypted ssh session ofc. As long as the call is not encrypted, everything is fine, I can call any user I want. But when I started to encrypt my traffic everything went wrong and I can't find why.I've generated certficate for both client and server, the traffic is encrypted because I can't see anything in wireshark(I can't see encrypted traffic but I see non-encrypted traffic). Blink is configured correctly with SDES mandatory, .pem file, car.crt , proxy on port 5061 tls, but I think the error is somewhere else. Myconig for sip.conf is like this: [general] udpbindaddr=0.0.0.0 tcpenable=yes ; Enable server for incoming TCP connections (default is no) tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces) ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060) tlsenable=yes ; Enable server for incoming TLS (secure) connections (default is no) tlsbindaddr=0.0.0.0 transport=udp disallow=all allow=ulaw ; Allow codecs in order of preference allow=alaw dtmfmode = rfc2833 tlscertfile=/etc/asterisk/keys/asterisk.pem tlscafile=/etc/asterisk/keys/ca.crt tlscipher=ALL tlsclientmethod=tlsv1 [201](can't communicate instant 488 error) type=friend username=vincent context=from-sip host=dynamic secret=not4usry callerid=vincent<201> mailbox=201#default nat=comedia transport=tls encryption=yes [203](no tls user and can communicate with the ones who don't use tls) type=friend username=antoine context=from-sip host=dynamic secret=not4usry callerid=antoine<203> mailbox=203#default nat=comedia Certificates have been generated using ./ast_tls_cert.... RTP logs == Using SIP RTP CoS mark 5 [May 11 15:34:33] WARNING[21893][C-00000d0e]: chan_sip.c:10803 process_sdp: Rejecting secure audio stream without encryption details: audio 50026 RTP/SAVP 113 9 0 8 101 SIP LOGS INVITE sip:203#vps466556.ovh.net SIP/2.0 Via: SIP/2.0/TLS 192.168.1.35:53076;rport;branch=z9hG4bKPj275105fe6a304b89b2b18ee5186b5085;alias Max-Forwards: 70 From: "Vincent" ;tag=523fe49e3a8646608481fbac0801b605 To: Contact: Call-ID: 0fb841de523e4ff0a74514247bb3445a CSeq: 4966 INVITE Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Supported: replaces, norefersub, gruu User-Agent: Blink 3.0.0 (Windows) Content-Type: application/sdp Content-Length: 425 v=0 o=- 3735043210 3735043210 IN IP4 192.168.1.35 s=Blink 3.0.0 (Windows) t=0 0 m=audio 50004 RTP/AVP 113 9 0 8 101 c=IN IP4 192.168.1.35 a=rtcp:50005 a=rtpmap:113 opus/48000/2 a=fmtp:113 useinbandfec=1 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=zrtp-hash:1.10 af10bf32a78e03147ffbf2859f96cc8d401048ee46a1f2cb961c20139b219913 a=sendrecv -- 2018-05-11 16:00:11.003276 [blink.exe 3320]: RECEIVED: Packet 136, +0:06:21.266013 54.37.8.124:5061 -(SIP over TLS)-> 192.168.1.35:53076 SIP/2.0 401 Unauthorized Via: SIP/2.0/TLS 192.168.1.35:53076;branch=z9hG4bKPj275105fe6a304b89b2b18ee5186b5085;alias;received=90.112.223.194;rport=53076 From: "Vincent" ;tag=523fe49e3a8646608481fbac0801b605 To: ;tag=as50eb1885 Call-ID: 0fb841de523e4ff0a74514247bb3445a CSeq: 4966 INVITE Server: Asterisk PBX 13.21.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer -Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6650a402" Content-Length: 0 -- 2018-05-11 16:00:11.004276 [blink.exe 3320]: SENDING: Packet 137, +0:06:21.267013 192.168.1.35:53076 -(SIP over TLS)-> 54.37.8.124:5061 ACK sip:203#vps466556.ovh.net SIP/2.0 Via: SIP/2.0/TLS 192.168.1.35:53076;rport;branch=z9hG4bKPj275105fe6a304b89b2b18ee5186b5085;alias Max-Forwards: 70 From: "Vincent" ;tag=523fe49e3a8646608481fbac0801b605 To: ;tag=as50eb1885 Call-ID: 0fb841de523e4ff0a74514247bb3445a CSeq: 4966 ACK User-Agent: Blink 3.0.0 (Windows) Content-Length: 0 -- 2018-05-11 16:00:11.005276 [blink.exe 3320]: SENDING: Packet 138, +0:06:21.268013 192.168.1.35:53076 -(SIP over TLS)-> 54.37.8.124:5061 INVITE sip:203#vps466556.ovh.net SIP/2.0 Via: SIP/2.0/TLS 192.168.1.35:53076;rport;branch=z9hG4bKPj3e8da342afaa41a385d9989648fd069f;alias Max-Forwards: 70 From: "Vincent" ;tag=523fe49e3a8646608481fbac0801b605 To: Contact: Call-ID: 0fb841de523e4ff0a74514247bb3445a CSeq: 4967 INVITE Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Supported: replaces, norefersub, gruu User-Agent: Blink 3.0.0 (Windows) Authorization: Digest username="201", realm="asterisk", nonce="6650a402", uri="sip:203#vps466556.ovh.net", response="dcd6fcd9d8b7381f86f07e1326aa9134", algorithm=MD5 Content-Type: application/sdp Content-Length: 425 v=0 o=- 3735043210 3735043210 IN IP4 192.168.1.35 s=Blink 3.0.0 (Windows) t=0 0 m=audio 50004 RTP/AVP 113 9 0 8 101 c=IN IP4 192.168.1.35 a=rtcp:50005 a=rtpmap:113 opus/48000/2 a=fmtp:113 useinbandfec=1 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=zrtp-hash:1.10 af10bf32a78e03147ffbf2859f96cc8d401048ee46a1f2cb961c20139b219913 a=sendrecv -- 2018-05-11 16:00:11.087226 [blink.exe 3320]: RECEIVED: Packet 139, +0:06:21.349963 54.37.8.124:5061 -(SIP over TLS)-> 192.168.1.35:53076 SIP/2.0 488 Not acceptable here Via: SIP/2.0/TLS 192.168.1.35:53076;branch=z9hG4bKPj3e8da342afaa41a385d9989648fd069f;alias;received=90.112.223.194;rport=53076 From: "Vincent" ;tag=523fe49e3a8646608481fbac0801b605 To: ;tag=as50eb1885 Call-ID: 0fb841de523e4ff0a74514247bb3445a CSeq: 4967 INVITE Server: Asterisk PBX 13.21.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 -- 2018-05-11 16:00:11.088227 [blink.exe 3320]: SENDING: Packet 140, +0:06:21.350964 192.168.1.35:53076 -(SIP over TLS)-> 54.37.8.124:5061 ACK sip:203#vps466556.ovh.net SIP/2.0 Via: SIP/2.0/TLS 192.168.1.35:53076;rport;branch=z9hG4bKPj3e8da342afaa41a385d9989648fd069f;alias Max-Forwards: 70 From: "Vincent" ;tag=523fe49e3a8646608481fbac0801b605 To: ;tag=as50eb1885 Call-ID: 0fb841de523e4ff0a74514247bb3445a CSeq: 4967 ACK User-Agent: Blink 3.0.0 (Windows) Content-Length: 0 Thanks for your help, Vince
Your SIP endpoint specifies encryption=yes, but the INVITE from your client specifies RTP/AVP, not RTP/SAVP. iirc, Blink has an option for optimistic or mandatory encryption; you'll need to change it to mandatory.
If you have the same problem as me, reinstall the Srtp library and asterisk, also log sip request because I had a bug where Blink didn't send register request anymore (even when pushing that register button 1000times) so I downloaded PhonerLite and everything worked perfectly
Can't connect to localhost after upgrading to El Capitan
After updating to El Capitan I can't access localhost. Another computer where El Capitan was already at the installation of MAMP everything works fine. Both Apache and MySQL server start ,but can't get to localhost page This is what I've tried so far: 1) Uncommented some of the lines on httpd.conf as indicated on some messages 2) Replaced current files with files in private/etc/apache2/original 3) Copied http.conf and extra/ from the other computer (the one where localhost is working) 4) Tried different ports configuration (Apache: 8888 & MySql: 8889) Since none of the above worked, I put back files to the original state (i.e. before uncommenting or replacing) Finally, if instead of localhost I type computer_name.local/ on the browser address bar I can see my files. I am VERY new with all this, any suggestions? Thanks, S Denis, thanks for the suggestion. These are the results from Terminal: lo0: flags=8049 mtu 16384 options=3 inet6 ::1 prefixlen 128 inet 127.0.0.1 netmask 0xff000000 inet6 fe80::1%lo0 prefixlen 64 scopeid 0x1 nd6 options=1 gif0: flags=8010 mtu 1280 stf0: flags=0<> mtu 1280 en1: flags=8823 mtu 1500 ether 88:63:df:cb:a5:39 nd6 options=1 media: autoselect () status: inactive en0: flags=8863 mtu 1500 options=10b ether ac:87:a3:14:72:1b nd6 options=1 media: autoselect (none) status: inactive en2: flags=963 mtu 1500 options=60 ether 0a:00:00:3c:bc:00 media: autoselect status: inactive en3: flags=963 mtu 1500 options=60 ether 0a:00:00:3c:bc:01 media: autoselect status: inactive en5: flags=8863 mtu 1500 options=10b ether 40:6c:8f:38:4f:a5 inet6 fe80::426c:8fff:fe38:4fa5%en5 prefixlen 64 scopeid 0x8 inet 192.168.1.21 netmask 0xffffff00 broadcast 192.168.1.255 nd6 options=1 media: autoselect (1000baseT ) status: active en7: flags=8863 mtu 1460 ether 02:50:f2:00:00:01 nd6 options=1 media: autoselect status: inactive p2p0: flags=8802 mtu 2304 ether 0a:63:df:cb:a5:39 media: autoselect status: inactive bridge0: flags=8863 mtu 1500 options=63 ether 8a:63:df:bc:63:00 Configuration: id 0:0:0:0:0:0 priority 0 hellotime 0 fwddelay 0 maxage 0 holdcnt 0 proto stp maxaddr 100 timeout 1200 root id 0:0:0:0:0:0 priority 0 ifcost 0 port 0 ipfilter disabled flags 0x2 member: en2 flags=3 ifmaxaddr 0 port 6 priority 0 path cost 0 member: en3 flags=3 ifmaxaddr 0 port 7 priority 0 path cost 0 nd6 options=1 media: status: inactive awdl0: flags=8902 mtu 1484 ether 52:4f:2b:35:96:9d nd6 options=1 media: autoselect status: inactive
How to remove unnecessary data from SIP Invite in sipML5
How can I remove unnecessary data from SIP invite in sipML5? Now it's too big when i sending it to my server (need only audio). It will accept maximum of 1,500 bytes and it must be on UDP. Could you tell me how to do it ? how to remove some codecs, etc ? I don't know anything about it, just learning now sipML. In other posts there are answers like remove codecs, but there is no answers how to do it :) My invite: SEND: INVITE sip:some_client#some_address SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKRyhpzJOIUVBDwgMLxDIq1CAmXFZo2HkD;rport From: <sip:some_client#some_address>;tag=hpGTFTQ0Kpt6JFgsn8Bc To: <sip:some_number_to_call#some_address> Contact: "undefined"<sip:some_client#df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws> Call-ID: 558d80b1-b383-344e-dea1-b95ddbe9dc3f CSeq: 30366 INVITE Content-Type: application/sdp Content-Length: 2247 Max-Forwards: 70 v=0 o=- 3717615351353762000 2 IN IP4 127.0.0.1 s=Doubango Telecom - chrome t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS HMFRvujnzsUIWdP6940nngmFOxrtVbMeG8nr m=audio 52548 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126 c=IN IP4 91.241.5.1 a=rtcp:11803 IN IP4 91.241.5.1 a=candidate:2002913928 1 udp 2122260223 192.168.81.65 52691 typ host generation 0 a=candidate:2002913928 2 udp 2122260222 192.168.81.65 58663 typ host generation 0 a=candidate:4129950780 1 udp 1686052607 91.241.5.1 52548 typ srflx raddr 192.168.81.65 rport 52691 generation 0 a=candidate:4129950780 2 udp 1686052606 91.241.5.1 11803 typ srflx raddr 192.168.81.65 rport 58663 generation 0 a=candidate:971110008 1 tcp 1518280447 192.168.81.65 9 typ host tcptype active generation 0 a=candidate:971110008 2 tcp 1518280446 192.168.81.65 9 typ host tcptype active generation 0 a=candidate:4129950780 1 udp 1686052607 91.241.5.1 44693 typ srflx raddr 192.168.81.65 rport 52691 generation 0 a=candidate:4129950780 2 udp 1686052606 91.241.5.1 47874 typ srflx raddr 192.168.81.65 rport 58663 generation 0 a=candidate:4129950780 1 udp 1686052607 91.241.5.1 22880 typ srflx raddr 192.168.81.65 rport 52691 generation 0 a=candidate:4129950780 2 udp 1686052606 91.241.5.1 19665 typ srflx raddr 192.168.81.65 rport 58663 generation 0 a=ice-ufrag:FOoiTf25RFgO/bOx a=ice-pwd:ANc8oBQwW5zwBMd9lK2slJNN a=fingerprint:sha-256 78:A2:A4:13:11:A2:74:25:6E:B8:D4:9E:F3:1B:71:7E:A5:10:38:39:01:CC:93:C1:74:B3:96:25:71:C8:D2:5D 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=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:1519941870 cname:s8rO7shwZbHh0wyw a=ssrc:1519941870 msid:HMFRvujnzsUIWdP6940nngmFOxrtVbMeG8nr 38192f79-8fcc-45c9-b812-55e01c26364e a=ssrc:1519941870 mslabel:HMFRvujnzsUIWdP6940nngmFOxrtVbMeG8nr a=ssrc:1519941870 label:38192f79-8fcc-45c9-b812-55e01c26364e
It is not recommended to modify this on the client side from JavaScript. Your server or WebRTC to SIP gateway should solve this automatically (by forwarding on TCP or removing unnecessary lines if can be forwarded on UDP only).