몇 주 동안 WebRTC용 Asterisk와 씨름한 후, 나는 이 포럼에 내 문제를 올리기로 결정했습니다. 내 문제는 다음과 같습니다.
WebRTC에서 WebRTC 클라이언트로 오디오를 가져오지 못하고 있습니다. 저는 LAN 환경에서 일하고 있습니다.
상황 - 독립 실행형 Asterisk 서버를 사용하여 JSSIP에서 JSSIP(동일 클라이언트)로 호출합니다.
문제
6001(JSSIP)에서 6002(JSSIP)로 통화시 소리가 전혀 나오지 않습니다. 잠시 후 일부 RTP 패킷이 전송되지만 수신되지 않습니다. RTP 패킷이 로그 파일에 표시된 후 약 40초가 지나면 사용자의 작업 없이 통화가 종료됩니다.
두 클라이언트 모두 동일한 컴퓨터에 있고 둘 다 Google 크롬에 있으며 하나는 시크릿 모드로 실행되고 있습니다. 또한 2개의 서로 다른 노트북으로 테스트했지만 결과는 동일했습니다.
이 문제의 원인을 아는 사람이 있나요? 나는 또한 다음 스레드를 따랐습니다.http://forums.digium.com/viewtopic.php?f=1&t=90167, 그러나 이것은 나에게 효과가 없었습니다. 아래에서 로그 파일과 추가 정보를 확인할 수 있습니다.
동일한 문제에 대한 여러 게시물로 인해 이 문제를 해결하려고 했습니다.
- 프록시 뒤에서 Asterisk 실행(oversip)
- 미디어 코딩을 켜고 끄는 webrtc2sip 사용
- RTCwebbreaker와 함께 SipML5 사용, 비디오 비활성화
- 별표를 공용으로 설정합니다(WAN 주소를 사용하고 서버로 포트 전달).
- CentOS에서 별표 실행
- 중고 파이어폭스
별표 서버:
- OS: 우분투 14.04
- 버전: 별표 13.2 및 FreePBX 12
- IP 주소: 192.168.178.203
- 웹소켓 포트: 8088
- SIP 포트: 5060
고객:
- 운영체제: 윈도우 8.1
- 브라우저: Chrome 버전 40.0.2214.115m
- IP 주소: 192.168.178.18
구성:
rtp.conf
[general]
rtpstart=10000
rtpend=20000
icesupport=yes
stunaddr=74.125.132.127:19302
http.conf
[general]
enabled=yes
enablestatic=no
bindaddr=0.0.0.0
bindport=8088
sip.conf
[general]
port=5060
bindaddr=192.168.178.203
transport=ws,wss,udp
[6001]
host=dynamic
secret=6001
context=internal
type=friend
transport=ws,wss,udp
encryption=yes
avpf=yes
force_avp=yes
icesupport=yes
directmedia=outgoing
disallow=all
allow=opus,ulaw,vp8
dtlsenable=yes
dtlsverify=fingerprint
dtlscertfile=/etc/asterisk/keys/default.pem
dtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass
nat=no
[6002]
host=dynamic
secret=6002
context=internal
type=friend
transport=ws,wss,udp
encryption=yes
avpf=yes
force_avp=yes
icesupport=yes
directmedia=outgoing
disallow=all
allow=opus,ulaw,vp8
dtlsenable=yes
dtlsverify=fingerprint
dtlscertfile=/etc/asterisk/keys/default.pem
dtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass
nat=no
확장.conf
[internal]
exten => 6001,1,Dial(SIP/6001)
exten => 6002,1,Dial(SIP/6002)
벌채 반출
전체 서버 로그는 여기에 있습니다:http://pastebin.com/qHwzXbbd
sip 및 rtp 디버깅이 활성화된 서버 로깅의 일부입니다.
<------------>
Scheduling destruction of SIP dialog '0vreoff3cjblcgubthf1df' in 32000 ms (Method: REGISTER)
<--- SIP read from WS:192.168.178.18:55387 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/WS 1t4mt0723j5t.invalid;branch=z9hG4bK7239492
Max-Forwards: 69
To: <sip:[email protected]:5060>
From: "6001" <sip:[email protected]:5060>;tag=lq9ascrlba
Call-ID: sf6m6uln2amn2lhuuvia
CSeq: 7253 INVITE
X-Can-Renegotiate: undefined
Contact: <sip:[email protected];transport=ws;ob>
Content-Type: application/sdp
Session-Expires: 90
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS
Supported: timer,ice,outbound
User-Agent: JsSIP 0.6.18
Content-Length: 4710
v=0
o=- 5530171873894353207 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio video
a=msid-semantic: WMS bEWKeIMVkeE6PTP0lgDPhtJoGfmy7MeXYx0v
m=audio 62227 RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
c=IN IP4 192.168.12.1
a=rtcp:62227 IN IP4 192.168.12.1
a=candidate:3708951358 1 udp 2122129151 192.168.12.1 62227 typ host generation 0
a=candidate:3708951358 2 udp 2122129151 192.168.12.1 62227 typ host generation 0
a=candidate:3242410012 1 udp 2122063615 25.124.74.215 62228 typ host generation 0
a=candidate:3242410012 2 udp 2122063615 25.124.74.215 62228 typ host generation 0
a=candidate:1867667642 1 udp 2121998079 192.168.254.1 62229 typ host generation 0
a=candidate:1867667642 2 udp 2121998079 192.168.254.1 62229 typ host generation 0
a=candidate:1340102981 1 udp 2121932543 192.168.178.18 62230 typ host generation 0
a=candidate:1340102981 2 udp 2121932543 192.168.178.18 62230 typ host generation 0
a=candidate:2475812814 1 tcp 1518149375 192.168.12.1 0 typ host tcptype active generation 0
a=candidate:2475812814 2 tcp 1518149375 192.168.12.1 0 typ host tcptype active generation 0
a=candidate:2411773164 1 tcp 1518083839 25.124.74.215 0 typ host tcptype active generation 0
a=candidate:2411773164 2 tcp 1518083839 25.124.74.215 0 typ host tcptype active generation 0
a=candidate:567387210 1 tcp 1518018303 192.168.254.1 0 typ host tcptype active generation 0
a=candidate:567387210 2 tcp 1518018303 192.168.254.1 0 typ host tcptype active generation 0
a=candidate:23307701 1 tcp 1517952767 192.168.178.18 0 typ host tcptype active generation 0
a=candidate:23307701 2 tcp 1517952767 192.168.178.18 0 typ host tcptype active generation 0
a=ice-ufrag:KUz2P70p6fRKTNhU
a=ice-pwd:uSSZZuV537KDmpbtSzusDKLK
a=ice-options:google-ice
a=fingerprint:sha-256 E8:85:6A:27:CF:E0:50:A7:4B:E5:4D:25:70:76:D4:F1:5D:8D:79:93:40:4B:2E:6B:40:1D:F0:35:B9:38:56:E3
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
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:831281522 cname:+qS2XmtyXWZX3uFI
a=ssrc:831281522 msid:bEWKeIMVkeE6PTP0lgDPhtJoGfmy7MeXYx0v dc973b27-9b15-4cdd-bbf8-80807c4d3b1d
a=ssrc:831281522 mslabel:bEWKeIMVkeE6PTP0lgDPhtJoGfmy7MeXYx0v
a=ssrc:831281522 label:dc973b27-9b15-4cdd-bbf8-80807c4d3b1d
m=video 62227 RTP/SAVPF 100 116 117 96
c=IN IP4 192.168.12.1
a=rtcp:62227 IN IP4 192.168.12.1
a=candidate:3708951358 1 udp 2122129151 192.168.12.1 62227 typ host generation 0
a=candidate:3708951358 2 udp 2122129151 192.168.12.1 62227 typ host generation 0
a=candidate:3242410012 1 udp 2122063615 25.124.74.215 62228 typ host generation 0
a=candidate:3242410012 2 udp 2122063615 25.124.74.215 62228 typ host generation 0
a=candidate:1867667642 1 udp 2121998079 192.168.254.1 62229 typ host generation 0
a=candidate:1867667642 2 udp 2121998079 192.168.254.1 62229 typ host generation 0
a=candidate:1340102981 1 udp 2121932543 192.168.178.18 62230 typ host generation 0
a=candidate:1340102981 2 udp 2121932543 192.168.178.18 62230 typ host generation 0
a=candidate:2475812814 1 tcp 1518149375 192.168.12.1 0 typ host tcptype active generation 0
a=candidate:2475812814 2 tcp 1518149375 192.168.12.1 0 typ host tcptype active generation 0
a=candidate:2411773164 1 tcp 1518083839 25.124.74.215 0 typ host tcptype active generation 0
a=candidate:2411773164 2 tcp 1518083839 25.124.74.215 0 typ host tcptype active generation 0
a=candidate:567387210 1 tcp 1518018303 192.168.254.1 0 typ host tcptype active generation 0
a=candidate:567387210 2 tcp 1518018303 192.168.254.1 0 typ host tcptype active generation 0
a=candidate:23307701 1 tcp 1517952767 192.168.178.18 0 typ host tcptype active generation 0
a=candidate:23307701 2 tcp 1517952767 192.168.178.18 0 typ host tcptype active generation 0
a=ice-ufrag:KUz2P70p6fRKTNhU
a=ice-pwd:uSSZZuV537KDmpbtSzusDKLK
a=ice-options:google-ice
a=fingerprint:sha-256 E8:85:6A:27:CF:E0:50:A7:4B:E5:4D:25:70:76:D4:F1:5D:8D:79:93:40:4B:2E:6B:40:1D:F0:35:B9:38:56:E3
a=setup:actpass
a=mid:video
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=recvonly
a=rtcp-mux
a=rtpmap:100 VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=rtcp-fb:100 goog-remb
a=rtpmap:116 red/90000
a=rtpmap:117 ulpfec/90000
a=rtpmap:96 rtx/90000
a=fmtp:96 apt=100
<------------->
--- (15 headers 89 lines) ---
Using INVITE request as basis request - sf6m6uln2amn2lhuuvia
Found peer '6001' for '6001' from 192.168.178.18:55387
<--- Reliably Transmitting (no NAT) to 192.168.178.18:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS 1t4mt0723j5t.invalid;branch=z9hG4bK7239492;received=192.168.178.18
From: "6001" <sip:[email protected]:5060>;tag=lq9ascrlba
To: <sip:[email protected]:5060>;tag=as6ed0dbe4
Call-ID: sf6m6uln2amn2lhuuvia
CSeq: 7253 INVITE
Server: FPBX-12.0.37(13.431072M)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="263201da"
Content-Length: 0
<------------>
10:57:50.828 utsx0x7f967800 ...STUN client transaction created
10:57:50.828 utsx0x7f967800 ...STUN sending message (transmit count=1)
10:57:50.828 icess0x7f970c0 ..Check 7: [2] 192.168.178.203:18141-->192.168.178.18:62230: state changed from Frozen to In Progress
10:57:50.839 utsx0x7f967800 STUN sending message (transmit count=2)
10:57:50.839 utsx0x7f967800 STUN sending message (transmit count=2)
10:57:50.850 icess0x7f970c0 Starting checklist periodic check
10:57:50.851 icess0x7f970c0 Starting checklist periodic check
10:57:50.861 utsx0x7f967800 STUN sending message (transmit count=2)
10:57:50.862 utsx0x7f967800 STUN sending message (transmit count=2)
10:57:53.770 stuse0x7f96fc0 ..TX 80 bytes STUN message to 192.168.178.18:62261:
--- begin STUN message ---
STUN Binding success response
Hdr: length=60, magic=2112a442, tsx_id=6d4d564652372f537652314a
Attributes:
XOR-MAPPED-ADDRESS: length=8, IPv4 addr=192.168.178.18:62261
SOFTWARE: length=12, value="pjnath-2.3.0"
MESSAGE-INTEGRITY: length=20, data=0204b9b98f375d27e045e50bd034f1ce99288e06
FINGERPRINT: length=4, value=1908015627 (0x71ba020b)
--- end of STUN message ---
10:57:55.586 utsx0x7f96f800 .....STUN client transaction created
10:57:55.586 utsx0x7f96f800 .....STUN sending message (transmit count=1)
10:57:55.586 icess0x7f96fc0 ....Check 7: [2] 192.168.178.203:13265-->192.168.178.18:62253: state changed from Frozen to In Progress
10:57:55.586 icess0x7f96fc0 .Performing delayed triggerred check for component 1
10:57:55.586 icess0x7f96fc0 ..Triggered check for check 6 not performed because it's in progress. Retransmitting
10:57:55.586 utsx0x7f96f801 ...STUN sending message (transmit count=1)
10:57:55.586 icess0x7f96fc0 .Performing delayed triggerred check for component 2
10:57:55.586 icess0x7f96fc0 ..Triggered check for check 7 not performed because it's in progress. Retransmitting
10:58:41.301 icess0x7f96fc0 .Triggered check for check 7 not performed because it's completed
10:58:41.301 icess0x7f96fc0 ..Check 7 is successful and nominated
Sent RTP packet to 192.168.178.18:62249 (via ICE) (type 00, seq 020194, ts 004320, len 000160)
Sent RTP packet to 192.168.178.18:62249 (via ICE) (type 00, seq 020195, ts 004480, len 000160)
10:58:41.509 stuse0x7f970c0 !Response cache deleted
10:58:41.509 stun_session.c tdata 0x7f96fc02cac8 destroy request, force=0, tsx=(nil)
Sent RTP packet to 192.168.178.18:62249 (via ICE) (type 00, seq 020204, ts 005920, len 000160)
10:58:41.519 stuse0x7f970c0 Response cache deleted
누군가 나에게 어떤 방향을 알려줄 수 있기를 바랍니다. 또는 누군가가 같은 문제에 직면하여 자신의 지식을 기꺼이 공유할 수 있기를 바랍니다.
자세한 내용은 저에게 문의해 주세요.
미리 감사드립니다.
답변1
구성 및 통화 설정이 괜찮아 보입니다. 200 OK 응답을 확인하는 것이 매우 중요합니다. 또한 브라우저에서 webrtc 로그를 활성화하고 이를 확인(또는 여기에서 제출)하여 실제로 (DTLS 인코딩된) SRTP를 보내는 위치를 확인할 수 있습니다.
에 대한 자세한 설명은 다음과 같습니다.Asterisk 13의 WebRTC 설정.
또한 기본적으로 SIP 신호에서 공용 IP를 감지하고 사용하려고 시도하기 때문에 공용 주소에서 별표를 실행하는 것이 좋습니다(또는 모든 개인 IP/공용 IP/NAT 구성을 다시 확인).