Asterisk 使用 SIPML5 和 MYSQL 接收代码 503
Asterisk receiving code 503 with SIPML5 and MYSQL
当我的用户在 sip.conf 文件中时，我确实使用 asterisk 11.16 和 SipML5 在两个浏览器之间进行了音频通话。
然后我在 sip.conf 中尝试了一个用户，在 MYSQL 中尝试了另一个用户，并且有效！所以在最后一步，MYSQL 中的两个客户端都不走运。
媒体请求只发生在一侧，一段时间后呼叫者收到 503 代码。
有什么线索吗？
星号：
== WebSocket connection from 'IP:PORT' for protocol 'sip' accepted using version '13'
== WebSocket connection from 'IP:PORT' for protocol 'sip' accepted using version '13'
-- Registered SIP '000001' at IP:PORT
> Saved useragent "IM-client/OMA1.0 sipML5-v1.0.0.0" for peer 000001
-- Registered SIP '002670' at IP:PORT
> Saved useragent "IM-client/OMA1.0 sipML5-v1.0.0.0" for peer 002670
== Using SIP RTP CoS mark 5
-- Executing [000001@from-internal:1] Dial("SIP/002670-00000002", "SIP/000001,60") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/000001
servidor-asterisk*CLI> core show channels
Channel Location State Application(Data)
SIP/002670-00000002 000001@from-internal Ring Dial(SIP/000001,60)
SIP/000001-00000003 000001@from-internal Down AppDial((Outgoing Line))
2 active channels
1 active call
2 calls processed
-- SIP/000001-00000003 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Auto fallthrough, channel 'SIP/002670-00000002' status is 'CONGESTION'
[Mar 20 16:08:48] WARNING[22762]: chan_sip.c:4020 retrans_pkt: Retransmission timeout reached on transmission 1c4882477bd3a6cd3541e672267d9356@IP:PORT for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
来电者：
"SIPML5 API version = 1.5.222" sip_api.js:1:16348
"Sistema -> Streaming nativo da webcam suportado. Iniciando camera..." IP:3001:556:3 Sistema -> Captura iniciada!" IP:3001:590:2
{id_interno: "2670", id_usuario_d: 1, nome: "Kim", sexo: "F", foto: "apres20150317174729.jpg", cidade: "Santa Cruz do Sul", favorito: false, red_nome: "Bapo", red_icone: "i_logo_bapo.png"} IP:3001:415
"User-Agent=Mozilla/5.0 (Windows NT 6.1; rv:36.0) Gecko/20100101 Firefox/36.0" sip_api.js:1:16348
"WebSocket supported = yes" sip_api.js:1:16348
"Navigator friendly name = firefox" sip_api.js:1:16348
"OS friendly name = windows" sip_api.js:1:16348
"Have WebRTC = yes" sip_api.js:1:16348
"Have GUM = yes" sip_api.js:1:16348
"Engine initialized" sip_api.js:1:16348
"s_websocket_server_url=ws://IP:8088/ws" sip_api.js:1:16348
"s_sip_outboundproxy_url=(null)" sip_api.js:1:16348
"b_rtcweb_breaker_enabled=no" sip_api.js:1:16348
"b_click2call_enabled=no" sip_api.js:1:16348
"b_early_ims=no" sip_api.js:1:16348
"b_enable_media_stream_cache=yes" sip_api.js:1:16348
"o_bandwidth={}" sip_api.js:1:16348
"o_video_size={}" sip_api.js:1:16348
"SIP stack start: proxy='ns313841.ovh.net:12060', realm='<sip:IP>', impi='002670', impu='"002670"<sip:002670@IP:8088>'" sip_api.js:1:16348
"Connecting to 'ws://IP:8088/ws'" sip_api.js:1:16348
"|> SipStack iniciado!" IP:3001:765:3
"__tsip_transport_ws_onopen" sip_api.js:1:16348
"State machine: tsip_dialog_register_Started_2_InProgress_X_oRegister" sip_api.js:1:16348
"SEND: REGISTER sip:IP SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK8HE4oAeuBcyOUbrE76PyBt4BptE9BsZ9;rport
From: "002670"<sip:002670@IP:8088>;tag=yd16vtKL2Ct5W8bgw8ea
To: "002670"<sip:002670@IP:8088>
Contact: "002670"<sip:002670@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=1800;click2call=no
Call-ID: fd274211-0331-7a38-7320-3fdb750d62c8
CSeq: 52764 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="002670",realm="IP",nonce="",uri="sip:IP",response=""
User-Agent: IM-client/OMA1.0 sipML5-v1.0.0.0
Organization: Tra La Lala Corp.
Supported: path
" sip_api.js:1:16348
"__tsip_transport_ws_onmessage" sip_api.js:1:16348
"recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=IP;branch=z9hG4bK8HE4oAeuBcyOUbrE76PyBt4BptE9BsZ9
From: "002670"<sip:002670@IP:8088>;tag=yd16vtKL2Ct5W8bgw8ea
To: "002670"<sip:002670@IP:8088>;tag=as67f4562a
Call-ID: fd274211-0331-7a38-7320-3fdb750d62c8
CSeq: 52764 REGISTER
Content-Length: 0
Server: Asterisk PBX 11.16.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
WWW-Authenticate: Digest realm="IP",nonce="4d3f211c",stale=FALSE,algorithm=MD5
" sip_api.js:1:16348
"State machine: tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494" sip_api.js:1:16348
"SEND: REGISTER sip:IP SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKtWK7SzDKHDcC7wMsgRxtM8tevHiZsSET;rport
From: "002670"<sip:002670@IP:8088>;tag=yd16vtKL2Ct5W8bgw8ea
To: "002670"<sip:002670@IP:8088>
Contact: "002670"<sip:002670@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=1800;click2call=no
Call-ID: fd274211-0331-7a38-7320-3fdb750d62c8
CSeq: 52765 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="002670",realm="IP",nonce="4d3f211c",uri="sip:IP",response="27379c933ca33f75ce31240d6385c254",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.0.0.0
Organization: Tra La Lala Corp.
Supported: path
" sip_api.js:1:16348
"State machine: c0000_Started_2_Outgoing_X_oINVITE" sip_api.js:1:16348
"PeerConnectionClass = function mozRTCPeerConnection() {
[native code]
} SessionDescriptionClass = function mozRTCSessionDescription() {
[native code]
} IceCandidateClass = function mozRTCIceCandidate() {
[native code]
}" sip_api.js:1:16348
"ICE servers:[{"url":"stun:stun.l.google.com:19302"}]" sip_api.js:1:16348
"__tsip_transport_ws_onmessage" sip_api.js:1:16348
"recv=SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=IP; branch=z9hG4bKtWK7SzDKHDcC7wMsgRxtM8tevHiZsSET
From: "002670"<sip:002670@IP:8088>;tag=yd16vtKL2Ct5W8bgw8ea
To: "002670"<sip:002670@IP:8088>;tag=as67f4562a
Contact: <sip:002670@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=1800
Call-ID: fd274211-0331-7a38-7320-3fdb750d62c8
CSeq: 52765 REGISTER
Expires: 1800
Content-Length: 0
Server: Asterisk PBX 11.16.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
Date: 20 Mar 2015 19:08:11 GMT;20
" sip_api.js:1:16348
"State machine: tsip_dialog_register_InProgress_2_Connected_X_2xx" sip_api.js:1:16348
"|> ELSE -> sipEventsListener: Media Requested" IP:3001:882:3
{type: "m_permission_requested", description: "Media Requested", o_event: Object} IP:3001:883
Mandatory/optional in createOffer options is deprecated! Use {"offerToReceiveAudio":true,"offerToReceiveVideo":false,"mozDontOfferDataChan nel":true} instead (note the case difference)! <desconhecido>
"onGetUserMediaSuccess" sip_api.js:1:16348
"createOffer" sip_api.js:1:16348
"|> ELSE -> sipEventsListener: Media Accepted" IP:3001:882:3
{type: "m_permission_accepted", description: "Media Accepted", o_event: Object} IP:3001:883
"|> ELSE -> sipEventsListener: Media Added" IP:3001:882:3
{type: "m_stream_audio_local_added", description: "Media Added", o_event: Object, session: Object} IP:3001:883
"onCreateSdpSuccess" sip_api.js:1:16348
"onSignalingstateChange:have-local-offer" sip_api.js:1:16348
"onSetLocalDescriptionSuccess" sip_api.js:1:16348
"onIceGatheringCompleted" sip_api.js:1:16348
"SEND: INVITE sip:000001@IP SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK0LJ9ofmfgrwoPW9WopzEIFHZIVTx6TgE;rport
From: "002670"<sip:002670@IP:8088>;tag=Sw9d0BOqG8wZCH8czu7B
To: <sip:000001@IP>
Contact: "002670"<sip:002670@df7jal23ls0d.invalid;rtcweb-breaker=no; click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr"
Call-ID: 4448657e-6b45-f1ae-2d45-417b49a19bd9
CSeq: 18237 INVITE
Content-Type: application/sdp
Content-Length: 1216
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.0.0.0
Organization: Tra La Lala Corp.
v=0
o=Mozilla-SIPUA-36.0.1 5071 0 IN IP4 0.0.0.0
s=Doubango Telecom - firefox
t=0 0
a=ice-ufrag:2969df08
a=ice-pwd:15fdd4182c02748e9d1a6ef9c2cbe979
a=fingerprint:sha-256 BB:C9:2E:56:BE:65:3D:4F:24:2A:AA:97:17:6F:C8:BB:8D:99:BA:5E:3E:F3:AD:D9:82:63:B7:8E:61:75:D0:BA
m=audio 9 UDP/TLS/RTP/SAVPF 109 9 0 8 101
c=IN IP4 0.0.0.0
a=rtpmap:109 opus/48000/2
a=ptime:20
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=setup:actpass
a=rtcp-mux
a=candidate:0 1 UDP 2128609535 192.168.1.113 56842 typ host
a=candidate:2 1 UDP 2128543999 192.168.1.104 56843 typ host
a=candidate:0 2 UDP 2128609534 192.168.1.113 56844 typ host
a=candidate:2 2 UDP 2128543998 192.168.1.104 56845 typ host
a=candidate:1 1 UDP 1692467199 IP 56842 typ srflx raddr 192.168.1.113 rport 56842
a=candidate:3 1 UDP 1692401663 IP 56843 typ srflx raddr 192.168.1.104 rport 56843
a=candidate:1 2 UDP 1692467198 IP 56844 typ srflx raddr 192.168.1.113 rport 56844
a=candidate:3 2 UDP 1692401662 IP 56845 typ srflx raddr 192.168.1.104 rport 56845
" sip_api.js:1:16348
"onIceCandidate = complete" sip_api.js:1:16348
"ICE GATHERING COMPLETED!" sip_api.js:1:16348
"onIceGatheringCompleted" sip_api.js:1:16348
"onIceGatheringCompleted but no local sdp request is pending" sip_api.js:1:16444
"onIceCandidate = complete" sip_api.js:1:16348
"ICE GATHERING COMPLETED!" sip_api.js:1:16348
"onIceGatheringCompleted" sip_api.js:1:16348
"onIceGatheringCompleted but no local sdp request is pending" sip_api.js:1:16444
"onIceCandidate = complete" sip_api.js:1:16348
"ICE GATHERING COMPLETED!" sip_api.js:1:16348
"onIceGatheringCompleted" sip_api.js:1:16348
"onIceGatheringCompleted but no local sdp request is pending" sip_api.js:1:16444
"onIceCandidate = complete" sip_api.js:1:16348
"ICE GATHERING COMPLETED!" sip_api.js:1:16348
"onIceGatheringCompleted" sip_api.js:1:16348
"onIceGatheringCompleted but no local sdp request is pending" sip_api.js:1:16444
"onIceCandidate = complete" sip_api.js:1:16348
"ICE GATHERING COMPLETED!" sip_api.js:1:16348
"onIceGatheringCompleted" sip_api.js:1:16348
"onIceGatheringCompleted but no local sdp request is pending" sip_api.js:1:16444
"onIceCandidate = complete" sip_api.js:1:16348
"ICE GATHERING COMPLETED!" sip_api.js:1:16348
"onIceGatheringCompleted" sip_api.js:1:16348
"onIceGatheringCompleted but no local sdp request is pending" sip_api.js:1:16444
"onIceCandidate = complete" sip_api.js:1:16348
"ICE GATHERING COMPLETED!" sip_api.js:1:16348
"onIceGatheringCompleted" sip_api.js:1:16348
"onIceGatheringCompleted but no local sdp request is pending" sip_api.js:1:16444
"onIceCandidate = complete" sip_api.js:1:16348
"ICE GATHERING COMPLETED!" sip_api.js:1:16348
"onIceGatheringCompleted" sip_api.js:1:16348
"onIceGatheringCompleted but no local sdp request is pending" sip_api.js:1:16444
"onIceCandidate = complete" sip_api.js:1:16348
"ICE GATHERING COMPLETED!" sip_api.js:1:16348
"onIceGatheringCompleted" sip_api.js:1:16348
"onIceGatheringCompleted but no local sdp request is pending" sip_api.js:1:16444
"__tsip_transport_ws_onmessage" sip_api.js:1:16348
"recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=IP; branch=z9hG4bK0LJ9ofmfgrwoPW9WopzEIFHZIVTx6TgE
From: "002670"<sip:002670@IP:8088>;tag=Sw9d0BOqG8wZCH8czu7B
To: <sip:000001@IP>;tag=as3720237e
Call-ID: 4448657e-6b45-f1ae-2d45-417b49a19bd9
CSeq: 18237 INVITE
Content-Length: 0
Server: Asterisk PBX 11.16.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
WWW-Authenticate: Digest realm="IP",nonce="18c6a5f8",stale=FALSE,algorithm=MD5
" sip_api.js:1:16348
"SEND: ACK sip:000001@IP SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK0LJ9ofmfgrwoPW9WopzEIFHZIVTx6TgE;rport
From: "002670"<sip:002670@IP:8088>;tag=Sw9d0BOqG8wZCH8czu7B
To: <sip:000001@IP>;tag=as3720237e
Call-ID: 4448657e-6b45-f1ae-2d45-417b49a19bd9
CSeq: 18237 ACK
Content-Length: 0
Max-Forwards: 70
" sip_api.js:1:16348
"State machine: x0000_Any_2_Any_X_i401_407_INVITE" sip_api.js:1:16348
"SEND: INVITE sip:000001@IP SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKBTtIE3wTk1dg8dAxX1IQl5p63SNt4rlV;rport
From: "002670"<sip:002670@IP:8088>;tag=Sw9d0BOqG8wZCH8czu7B
To: <sip:000001@IP>
Contact: "002670"<sip:002670@df7jal23ls0d.invalid;rtcweb-breaker=no; click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr"
Call-ID: 4448657e-6b45-f1ae-2d45-417b49a19bd9
CSeq: 18238 INVITE
Content-Type: application/sdp
Content-Length: 1216
Max-Forwards: 70
Authorization: Digest username="002670",realm="IP",nonce="18c6a5f8",uri="sip:000001@IP",response="07d31757427a917482a68ae9de6d739d",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.0.0.0
Organization: Tra La Lala Corp.
v=0
o=Mozilla-SIPUA-36.0.1 5071 0 IN IP4 0.0.0.0
s=Doubango Telecom - firefox
t=0 0
a=ice-ufrag:2969df08
a=ice-pwd:15fdd4182c02748e9d1a6ef9c2cbe979
a=fingerprint:sha-256 BB:C9:2E:56:BE:65:3D:4F:24:2A:AA:97:17:6F:C8:BB:8D:99:BA:5E:3E:F3:AD:D9:82:63 :B7:8E:61:75:D0:BA
m=audio 9 UDP/TLS/RTP/SAVPF 109 9 0 8 101
c=IN IP4 0.0.0.0
a=rtpmap:109 opus/48000/2
a=ptime:20
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=setup:actpass
a=rtcp-mux
a=candidate:0 1 UDP 2128609535 192.168.1.113 56842 typ host
a=candidate:2 1 UDP 2128543999 192.168.1.104 56843 typ host
a=candidate:0 2 UDP 2128609534 192.168.1.113 56844 typ host
a=candidate:2 2 UDP 2128543998 192.168.1.104 56845 typ host
=candidate:1 1 UDP 1692467199 IP 56842 typ srflx raddr 192.168.1.113 rport 56842
a=candidate:3 1 UDP 1692401663 IP 56843 typ srflx raddr 192.168.1.104 rport 56843
a=candidate:1 2 UDP 1692467198 IP 56844 typ srflx raddr 192.168.1.113 rport 56844
a=candidate:3 2 UDP 1692401662 IP 56845 typ srflx raddr 192.168.1.104 rport 56845
" sip_api.js:1:16348
"__tsip_transport_ws_onmessage" sip_api.js:1:16348
"recv=SIP/2.0 100 Trying
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=IP;branch=z9hG4bKBTtIE3wTk1dg8dAxX1IQl5p63SNt4rlV
From: "002670"<sip:002670@IP:8088>;tag=Sw9d0BOqG8wZCH8czu7B
To: <sip:000001@IP>
Contact: <sip:000001@IP:5060;transport=WS>
Call-ID: 4448657e-6b45-f1ae-2d45-417b49a19bd9
CSeq: 18238 INVITE
Content-Length: 0
Server: Asterisk PBX 11.16.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
" sip_api.js:1:16348
"State machine: x0000_Any_2_Any_X_i1xx" sip_api.js:1:16348
"__tsip_transport_ws_onmessage" sip_api.js:1:16348
"recv=SIP/2.0 503 Service Unavailable
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=IP;branch=z9hG4bKBTtIE3wTk1dg8dAxX1IQl5p63SNt4rlV
From: "002670"<sip:002670@IP:8088>;tag=Sw9d0BOqG8wZCH8czu7B
To: <sip:000001@IP>;tag=as4f8494c0
Call-ID: 4448657e-6b45-f1ae-2d45-417b49a19bd9
CSeq: 18238 INVITE
Content-Length: 0
Server: Asterisk PBX 11.16.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
" sip_api.js:1:16348
"SEND: ACK sip:000001@IP SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKBTtIE3wTk1dg8dAxX1IQl5p63SNt4rlV;rport
From: "002670"<sip:002670@IP:8088>;tag=Sw9d0BOqG8wZCH8czu7B
To: <sip:000001@IP>;tag=as4f8494c0
Call-ID: 4448657e-6b45-f1ae-2d45-417b49a19bd9
CSeq: 18238 ACK
Content-Length: 0
Max-Forwards: 70
" sip_api.js:1:16348
"State machine: c0000_Outgoing_2_Terminated_X_i300_to_i699INVITE" sip_api.js:1:16348
"=== INVITE Dialog terminated ===" sip_api.js:1:16348
"onSignalingstateChange:closed" sip_api.js:1:16348
"PeerConnection::stop()" sip_api.js:1:16348
"The FSM is in the final state" sip_api.js:1:16444
呼叫：
"SIPML5 API version = 1.5.222" sip_api.js:1:16348
"Sistema -> Streaming nativo da webcam suportado. Iniciando camera..." IP:3001:556:3
"Sistema -> Captura iniciada!" IP:3001:590:2
"User-Agent=Mozilla/5.0 (Windows NT 6.1; rv:36.0) Gecko/20100101 Firefox/36.0" sip_api.js:1:16348
"WebSocket supported = yes" sip_api.js:1:16348
"Navigator friendly name = firefox" sip_api.js:1:16348
"OS friendly name = windows" sip_api.js:1:16348
"Have WebRTC = yes" sip_api.js:1:16348
"Have GUM = yes" sip_api.js:1:16348
"Engine initialized" sip_api.js:1:16348
"s_websocket_server_url=ws://IP:8088/ws" sip_api.js:1:16348
"s_sip_outboundproxy_url=(null)" sip_api.js:1:16348
"b_rtcweb_breaker_enabled=no" sip_api.js:1:16348
"b_click2call_enabled=no" sip_api.js:1:16348
"b_early_ims=no" sip_api.js:1:16348
"b_enable_media_stream_cache=yes" sip_api.js:1:16348
"o_bandwidth={}" sip_api.js:1:16348
"o_video_size={}" sip_api.js:1:16348
"SIP stack start: proxy='ns313841.ovh.net:10060', realm='<sip:IP>', impi='000001', impu='"000001"<sip:000001@IP:8088>'" sip_api.js:1:16348
"Connecting to 'ws://IP:8088/ws'" sip_api.js:1:16348
"|> SipStack iniciado!" IP:3001:765:3
"__tsip_transport_ws_onopen" sip_api.js:1:16348
{id_interno: "1", id_usuario_d: 2670, nome: "Coisa", sexo: "M", foto: "Coisa20150317152721.jpg", cidade: "Santa Cruz do Sul", favorito: true, red_nome: "Bapo", red_icone: "i_logo_bapo.png"} IP:3001:415
"State machine: tsip_dialog_register_Started_2_InProgress_X_oRegister" sip_api.js:1:16348
"SEND: REGISTER sip:IP SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK2zpqnfk8SZSCtFZJhDfT3xeMOrJhkTuX;rport
From: "000001"<sip:000001@IP:8088>;tag=21sD3hs1CpOqrOBO9NsX
To: "000001"<sip:000001@IP:8088>
Contact: "000001"<sip:000001@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=1800;click2call=no
Call-ID: 812515ca-07e6-98e7-9892-f18a83d2aa30
CSeq: 22917 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="000001",realm="IP",nonce="",uri="sip:IP",response=""
User-Agent: IM-client/OMA1.0 sipML5-v1.0.0.0
Organization: Tra La Lala Corp.
Supported: path
" sip_api.js:1:16348
"__tsip_transport_ws_onmessage" sip_api.js:1:16348
"recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=IP; branch=z9hG4bK2zpqnfk8SZSCtFZJhDfT3xeMOrJhkTuX
From: "000001"<sip:000001@IP:8088>;tag=21sD3hs1CpOqrOBO9NsX
To: "000001"<sip:000001@IP:8088>;tag=as427ae015
Call-ID: 812515ca-07e6-98e7-9892-f18a83d2aa30
CSeq: 22917 REGISTER
Content-Length: 0
Server: Asterisk PBX 11.16.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
WWW-Authenticate: Digest realm="IP",nonce="7931c3b8",stale=FALSE,algorithm=MD5
" sip_api.js:1:16348
"State machine: tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494" sip_api.js:1:16348
"SEND: REGISTER sip:IP SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKEXbpaIGIxGtPKm1WPqUtFLnqSlfCNRZB;rport
From: "000001"<sip:000001@IP:8088>;tag=21sD3hs1CpOqrOBO9NsX
To: "000001"<sip:000001@IP:8088>
Contact: "000001"<sip:000001@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=1800;click2call=no
Call-ID: 812515ca-07e6-98e7-9892-f18a83d2aa30
CSeq: 22918 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="000001",realm="IP",nonce="7931c3b8",uri="sip:IP",response="2bd91f75bef84bf128996489ba8b743f",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.0.0.0
Organization: Tra La Lala Corp.
Supported: path
" sip_api.js:1:16348
"__tsip_transport_ws_onmessage" sip_api.js:1:16348
"recv=SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=IP; branch=z9hG4bKEXbpaIGIxGtPKm1WPqUtFLnqSlfCNRZB
From: "000001"<sip:000001@IP:8088>;tag=21sD3hs1CpOqrOBO9NsX
To: "000001"<sip:000001@IP:8088>;tag=as427ae015
Contact: <sip:000001@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=1800
Call-ID: 812515ca-07e6-98e7-9892-f18a83d2aa30
CSeq: 22918 REGISTER
Expires: 1800
Content-Length: 0
Server: Asterisk PBX 11.16.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
Date: 20 Mar 2015 19:08:10 GMT;20
" sip_api.js:1:16348
"State machine: tsip_dialog_register_InProgress_2_Connected_X_2xx"`
所以它正在工作，对于可能遇到此问题的任何其他人，sip.conf 文件缺少一行：
[general]
udpbindaddr=0.0.0.0:5060
realm=IP ;replace with your Asterisk server public IP address or host
rtcachefriends=yes
transport=udp,ws,wss
videosupport=no
avpf=yes
icesupport=yes
directmedia=no
allowguest=no
allwaysreject=yes
rtptimeout=30
allow=all
'rtcachefriends=yes' 似乎是实时所必需的。
在我的 mysql table 中，'qualify' 字段默认为 NULL，应该是。
这就是所有人！
当我的用户在 sip.conf 文件中时，我确实使用 asterisk 11.16 和 SipML5 在两个浏览器之间进行了音频通话。
然后我在 sip.conf 中尝试了一个用户，在 MYSQL 中尝试了另一个用户，并且有效！所以在最后一步，MYSQL 中的两个客户端都不走运。
媒体请求只发生在一侧，一段时间后呼叫者收到 503 代码。 有什么线索吗？
星号：
== WebSocket connection from 'IP:PORT' for protocol 'sip' accepted using version '13'
== WebSocket connection from 'IP:PORT' for protocol 'sip' accepted using version '13'
-- Registered SIP '000001' at IP:PORT
> Saved useragent "IM-client/OMA1.0 sipML5-v1.0.0.0" for peer 000001
-- Registered SIP '002670' at IP:PORT
> Saved useragent "IM-client/OMA1.0 sipML5-v1.0.0.0" for peer 002670
== Using SIP RTP CoS mark 5
-- Executing [000001@from-internal:1] Dial("SIP/002670-00000002", "SIP/000001,60") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/000001
servidor-asterisk*CLI> core show channels
Channel Location State Application(Data)
SIP/002670-00000002 000001@from-internal Ring Dial(SIP/000001,60)
SIP/000001-00000003 000001@from-internal Down AppDial((Outgoing Line))
2 active channels
1 active call
2 calls processed
-- SIP/000001-00000003 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Auto fallthrough, channel 'SIP/002670-00000002' status is 'CONGESTION'
[Mar 20 16:08:48] WARNING[22762]: chan_sip.c:4020 retrans_pkt: Retransmission timeout reached on transmission 1c4882477bd3a6cd3541e672267d9356@IP:PORT for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
来电者：
"SIPML5 API version = 1.5.222" sip_api.js:1:16348
"Sistema -> Streaming nativo da webcam suportado. Iniciando camera..." IP:3001:556:3 Sistema -> Captura iniciada!" IP:3001:590:2
{id_interno: "2670", id_usuario_d: 1, nome: "Kim", sexo: "F", foto: "apres20150317174729.jpg", cidade: "Santa Cruz do Sul", favorito: false, red_nome: "Bapo", red_icone: "i_logo_bapo.png"} IP:3001:415
"User-Agent=Mozilla/5.0 (Windows NT 6.1; rv:36.0) Gecko/20100101 Firefox/36.0" sip_api.js:1:16348
"WebSocket supported = yes" sip_api.js:1:16348
"Navigator friendly name = firefox" sip_api.js:1:16348
"OS friendly name = windows" sip_api.js:1:16348
"Have WebRTC = yes" sip_api.js:1:16348
"Have GUM = yes" sip_api.js:1:16348
"Engine initialized" sip_api.js:1:16348
"s_websocket_server_url=ws://IP:8088/ws" sip_api.js:1:16348
"s_sip_outboundproxy_url=(null)" sip_api.js:1:16348
"b_rtcweb_breaker_enabled=no" sip_api.js:1:16348
"b_click2call_enabled=no" sip_api.js:1:16348
"b_early_ims=no" sip_api.js:1:16348
"b_enable_media_stream_cache=yes" sip_api.js:1:16348
"o_bandwidth={}" sip_api.js:1:16348
"o_video_size={}" sip_api.js:1:16348
"SIP stack start: proxy='ns313841.ovh.net:12060', realm='<sip:IP>', impi='002670', impu='"002670"<sip:002670@IP:8088>'" sip_api.js:1:16348
"Connecting to 'ws://IP:8088/ws'" sip_api.js:1:16348
"|> SipStack iniciado!" IP:3001:765:3
"__tsip_transport_ws_onopen" sip_api.js:1:16348
"State machine: tsip_dialog_register_Started_2_InProgress_X_oRegister" sip_api.js:1:16348
"SEND: REGISTER sip:IP SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK8HE4oAeuBcyOUbrE76PyBt4BptE9BsZ9;rport
From: "002670"<sip:002670@IP:8088>;tag=yd16vtKL2Ct5W8bgw8ea
To: "002670"<sip:002670@IP:8088>
Contact: "002670"<sip:002670@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=1800;click2call=no
Call-ID: fd274211-0331-7a38-7320-3fdb750d62c8
CSeq: 52764 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="002670",realm="IP",nonce="",uri="sip:IP",response=""
User-Agent: IM-client/OMA1.0 sipML5-v1.0.0.0
Organization: Tra La Lala Corp.
Supported: path
" sip_api.js:1:16348
"__tsip_transport_ws_onmessage" sip_api.js:1:16348
"recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=IP;branch=z9hG4bK8HE4oAeuBcyOUbrE76PyBt4BptE9BsZ9
From: "002670"<sip:002670@IP:8088>;tag=yd16vtKL2Ct5W8bgw8ea
To: "002670"<sip:002670@IP:8088>;tag=as67f4562a
Call-ID: fd274211-0331-7a38-7320-3fdb750d62c8
CSeq: 52764 REGISTER
Content-Length: 0
Server: Asterisk PBX 11.16.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
WWW-Authenticate: Digest realm="IP",nonce="4d3f211c",stale=FALSE,algorithm=MD5
" sip_api.js:1:16348
"State machine: tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494" sip_api.js:1:16348
"SEND: REGISTER sip:IP SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKtWK7SzDKHDcC7wMsgRxtM8tevHiZsSET;rport
From: "002670"<sip:002670@IP:8088>;tag=yd16vtKL2Ct5W8bgw8ea
To: "002670"<sip:002670@IP:8088>
Contact: "002670"<sip:002670@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=1800;click2call=no
Call-ID: fd274211-0331-7a38-7320-3fdb750d62c8
CSeq: 52765 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="002670",realm="IP",nonce="4d3f211c",uri="sip:IP",response="27379c933ca33f75ce31240d6385c254",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.0.0.0
Organization: Tra La Lala Corp.
Supported: path
" sip_api.js:1:16348
"State machine: c0000_Started_2_Outgoing_X_oINVITE" sip_api.js:1:16348
"PeerConnectionClass = function mozRTCPeerConnection() {
[native code]
} SessionDescriptionClass = function mozRTCSessionDescription() {
[native code]
} IceCandidateClass = function mozRTCIceCandidate() {
[native code]
}" sip_api.js:1:16348
"ICE servers:[{"url":"stun:stun.l.google.com:19302"}]" sip_api.js:1:16348
"__tsip_transport_ws_onmessage" sip_api.js:1:16348
"recv=SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=IP; branch=z9hG4bKtWK7SzDKHDcC7wMsgRxtM8tevHiZsSET
From: "002670"<sip:002670@IP:8088>;tag=yd16vtKL2Ct5W8bgw8ea
To: "002670"<sip:002670@IP:8088>;tag=as67f4562a
Contact: <sip:002670@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=1800
Call-ID: fd274211-0331-7a38-7320-3fdb750d62c8
CSeq: 52765 REGISTER
Expires: 1800
Content-Length: 0
Server: Asterisk PBX 11.16.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
Date: 20 Mar 2015 19:08:11 GMT;20
" sip_api.js:1:16348
"State machine: tsip_dialog_register_InProgress_2_Connected_X_2xx" sip_api.js:1:16348
"|> ELSE -> sipEventsListener: Media Requested" IP:3001:882:3
{type: "m_permission_requested", description: "Media Requested", o_event: Object} IP:3001:883
Mandatory/optional in createOffer options is deprecated! Use {"offerToReceiveAudio":true,"offerToReceiveVideo":false,"mozDontOfferDataChan nel":true} instead (note the case difference)! <desconhecido>
"onGetUserMediaSuccess" sip_api.js:1:16348
"createOffer" sip_api.js:1:16348
"|> ELSE -> sipEventsListener: Media Accepted" IP:3001:882:3
{type: "m_permission_accepted", description: "Media Accepted", o_event: Object} IP:3001:883
"|> ELSE -> sipEventsListener: Media Added" IP:3001:882:3
{type: "m_stream_audio_local_added", description: "Media Added", o_event: Object, session: Object} IP:3001:883
"onCreateSdpSuccess" sip_api.js:1:16348
"onSignalingstateChange:have-local-offer" sip_api.js:1:16348
"onSetLocalDescriptionSuccess" sip_api.js:1:16348
"onIceGatheringCompleted" sip_api.js:1:16348
"SEND: INVITE sip:000001@IP SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK0LJ9ofmfgrwoPW9WopzEIFHZIVTx6TgE;rport
From: "002670"<sip:002670@IP:8088>;tag=Sw9d0BOqG8wZCH8czu7B
To: <sip:000001@IP>
Contact: "002670"<sip:002670@df7jal23ls0d.invalid;rtcweb-breaker=no; click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr"
Call-ID: 4448657e-6b45-f1ae-2d45-417b49a19bd9
CSeq: 18237 INVITE
Content-Type: application/sdp
Content-Length: 1216
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.0.0.0
Organization: Tra La Lala Corp.
v=0
o=Mozilla-SIPUA-36.0.1 5071 0 IN IP4 0.0.0.0
s=Doubango Telecom - firefox
t=0 0
a=ice-ufrag:2969df08
a=ice-pwd:15fdd4182c02748e9d1a6ef9c2cbe979
a=fingerprint:sha-256 BB:C9:2E:56:BE:65:3D:4F:24:2A:AA:97:17:6F:C8:BB:8D:99:BA:5E:3E:F3:AD:D9:82:63:B7:8E:61:75:D0:BA
m=audio 9 UDP/TLS/RTP/SAVPF 109 9 0 8 101
c=IN IP4 0.0.0.0
a=rtpmap:109 opus/48000/2
a=ptime:20
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=setup:actpass
a=rtcp-mux
a=candidate:0 1 UDP 2128609535 192.168.1.113 56842 typ host
a=candidate:2 1 UDP 2128543999 192.168.1.104 56843 typ host
a=candidate:0 2 UDP 2128609534 192.168.1.113 56844 typ host
a=candidate:2 2 UDP 2128543998 192.168.1.104 56845 typ host
a=candidate:1 1 UDP 1692467199 IP 56842 typ srflx raddr 192.168.1.113 rport 56842
a=candidate:3 1 UDP 1692401663 IP 56843 typ srflx raddr 192.168.1.104 rport 56843
a=candidate:1 2 UDP 1692467198 IP 56844 typ srflx raddr 192.168.1.113 rport 56844
a=candidate:3 2 UDP 1692401662 IP 56845 typ srflx raddr 192.168.1.104 rport 56845
" sip_api.js:1:16348
"onIceCandidate = complete" sip_api.js:1:16348
"ICE GATHERING COMPLETED!" sip_api.js:1:16348
"onIceGatheringCompleted" sip_api.js:1:16348
"onIceGatheringCompleted but no local sdp request is pending" sip_api.js:1:16444
"onIceCandidate = complete" sip_api.js:1:16348
"ICE GATHERING COMPLETED!" sip_api.js:1:16348
"onIceGatheringCompleted" sip_api.js:1:16348
"onIceGatheringCompleted but no local sdp request is pending" sip_api.js:1:16444
"onIceCandidate = complete" sip_api.js:1:16348
"ICE GATHERING COMPLETED!" sip_api.js:1:16348
"onIceGatheringCompleted" sip_api.js:1:16348
"onIceGatheringCompleted but no local sdp request is pending" sip_api.js:1:16444
"onIceCandidate = complete" sip_api.js:1:16348
"ICE GATHERING COMPLETED!" sip_api.js:1:16348
"onIceGatheringCompleted" sip_api.js:1:16348
"onIceGatheringCompleted but no local sdp request is pending" sip_api.js:1:16444
"onIceCandidate = complete" sip_api.js:1:16348
"ICE GATHERING COMPLETED!" sip_api.js:1:16348
"onIceGatheringCompleted" sip_api.js:1:16348
"onIceGatheringCompleted but no local sdp request is pending" sip_api.js:1:16444
"onIceCandidate = complete" sip_api.js:1:16348
"ICE GATHERING COMPLETED!" sip_api.js:1:16348
"onIceGatheringCompleted" sip_api.js:1:16348
"onIceGatheringCompleted but no local sdp request is pending" sip_api.js:1:16444
"onIceCandidate = complete" sip_api.js:1:16348
"ICE GATHERING COMPLETED!" sip_api.js:1:16348
"onIceGatheringCompleted" sip_api.js:1:16348
"onIceGatheringCompleted but no local sdp request is pending" sip_api.js:1:16444
"onIceCandidate = complete" sip_api.js:1:16348
"ICE GATHERING COMPLETED!" sip_api.js:1:16348
"onIceGatheringCompleted" sip_api.js:1:16348
"onIceGatheringCompleted but no local sdp request is pending" sip_api.js:1:16444
"onIceCandidate = complete" sip_api.js:1:16348
"ICE GATHERING COMPLETED!" sip_api.js:1:16348
"onIceGatheringCompleted" sip_api.js:1:16348
"onIceGatheringCompleted but no local sdp request is pending" sip_api.js:1:16444
"__tsip_transport_ws_onmessage" sip_api.js:1:16348
"recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=IP; branch=z9hG4bK0LJ9ofmfgrwoPW9WopzEIFHZIVTx6TgE
From: "002670"<sip:002670@IP:8088>;tag=Sw9d0BOqG8wZCH8czu7B
To: <sip:000001@IP>;tag=as3720237e
Call-ID: 4448657e-6b45-f1ae-2d45-417b49a19bd9
CSeq: 18237 INVITE
Content-Length: 0
Server: Asterisk PBX 11.16.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
WWW-Authenticate: Digest realm="IP",nonce="18c6a5f8",stale=FALSE,algorithm=MD5
" sip_api.js:1:16348
"SEND: ACK sip:000001@IP SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK0LJ9ofmfgrwoPW9WopzEIFHZIVTx6TgE;rport
From: "002670"<sip:002670@IP:8088>;tag=Sw9d0BOqG8wZCH8czu7B
To: <sip:000001@IP>;tag=as3720237e
Call-ID: 4448657e-6b45-f1ae-2d45-417b49a19bd9
CSeq: 18237 ACK
Content-Length: 0
Max-Forwards: 70
" sip_api.js:1:16348
"State machine: x0000_Any_2_Any_X_i401_407_INVITE" sip_api.js:1:16348
"SEND: INVITE sip:000001@IP SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKBTtIE3wTk1dg8dAxX1IQl5p63SNt4rlV;rport
From: "002670"<sip:002670@IP:8088>;tag=Sw9d0BOqG8wZCH8czu7B
To: <sip:000001@IP>
Contact: "002670"<sip:002670@df7jal23ls0d.invalid;rtcweb-breaker=no; click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr"
Call-ID: 4448657e-6b45-f1ae-2d45-417b49a19bd9
CSeq: 18238 INVITE
Content-Type: application/sdp
Content-Length: 1216
Max-Forwards: 70
Authorization: Digest username="002670",realm="IP",nonce="18c6a5f8",uri="sip:000001@IP",response="07d31757427a917482a68ae9de6d739d",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.0.0.0
Organization: Tra La Lala Corp.
v=0
o=Mozilla-SIPUA-36.0.1 5071 0 IN IP4 0.0.0.0
s=Doubango Telecom - firefox
t=0 0
a=ice-ufrag:2969df08
a=ice-pwd:15fdd4182c02748e9d1a6ef9c2cbe979
a=fingerprint:sha-256 BB:C9:2E:56:BE:65:3D:4F:24:2A:AA:97:17:6F:C8:BB:8D:99:BA:5E:3E:F3:AD:D9:82:63 :B7:8E:61:75:D0:BA
m=audio 9 UDP/TLS/RTP/SAVPF 109 9 0 8 101
c=IN IP4 0.0.0.0
a=rtpmap:109 opus/48000/2
a=ptime:20
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=setup:actpass
a=rtcp-mux
a=candidate:0 1 UDP 2128609535 192.168.1.113 56842 typ host
a=candidate:2 1 UDP 2128543999 192.168.1.104 56843 typ host
a=candidate:0 2 UDP 2128609534 192.168.1.113 56844 typ host
a=candidate:2 2 UDP 2128543998 192.168.1.104 56845 typ host
=candidate:1 1 UDP 1692467199 IP 56842 typ srflx raddr 192.168.1.113 rport 56842
a=candidate:3 1 UDP 1692401663 IP 56843 typ srflx raddr 192.168.1.104 rport 56843
a=candidate:1 2 UDP 1692467198 IP 56844 typ srflx raddr 192.168.1.113 rport 56844
a=candidate:3 2 UDP 1692401662 IP 56845 typ srflx raddr 192.168.1.104 rport 56845
" sip_api.js:1:16348
"__tsip_transport_ws_onmessage" sip_api.js:1:16348
"recv=SIP/2.0 100 Trying
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=IP;branch=z9hG4bKBTtIE3wTk1dg8dAxX1IQl5p63SNt4rlV
From: "002670"<sip:002670@IP:8088>;tag=Sw9d0BOqG8wZCH8czu7B
To: <sip:000001@IP>
Contact: <sip:000001@IP:5060;transport=WS>
Call-ID: 4448657e-6b45-f1ae-2d45-417b49a19bd9
CSeq: 18238 INVITE
Content-Length: 0
Server: Asterisk PBX 11.16.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
" sip_api.js:1:16348
"State machine: x0000_Any_2_Any_X_i1xx" sip_api.js:1:16348
"__tsip_transport_ws_onmessage" sip_api.js:1:16348
"recv=SIP/2.0 503 Service Unavailable
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=IP;branch=z9hG4bKBTtIE3wTk1dg8dAxX1IQl5p63SNt4rlV
From: "002670"<sip:002670@IP:8088>;tag=Sw9d0BOqG8wZCH8czu7B
To: <sip:000001@IP>;tag=as4f8494c0
Call-ID: 4448657e-6b45-f1ae-2d45-417b49a19bd9
CSeq: 18238 INVITE
Content-Length: 0
Server: Asterisk PBX 11.16.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
" sip_api.js:1:16348
"SEND: ACK sip:000001@IP SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKBTtIE3wTk1dg8dAxX1IQl5p63SNt4rlV;rport
From: "002670"<sip:002670@IP:8088>;tag=Sw9d0BOqG8wZCH8czu7B
To: <sip:000001@IP>;tag=as4f8494c0
Call-ID: 4448657e-6b45-f1ae-2d45-417b49a19bd9
CSeq: 18238 ACK
Content-Length: 0
Max-Forwards: 70
" sip_api.js:1:16348
"State machine: c0000_Outgoing_2_Terminated_X_i300_to_i699INVITE" sip_api.js:1:16348
"=== INVITE Dialog terminated ===" sip_api.js:1:16348
"onSignalingstateChange:closed" sip_api.js:1:16348
"PeerConnection::stop()" sip_api.js:1:16348
"The FSM is in the final state" sip_api.js:1:16444
呼叫：
"SIPML5 API version = 1.5.222" sip_api.js:1:16348
"Sistema -> Streaming nativo da webcam suportado. Iniciando camera..." IP:3001:556:3
"Sistema -> Captura iniciada!" IP:3001:590:2
"User-Agent=Mozilla/5.0 (Windows NT 6.1; rv:36.0) Gecko/20100101 Firefox/36.0" sip_api.js:1:16348
"WebSocket supported = yes" sip_api.js:1:16348
"Navigator friendly name = firefox" sip_api.js:1:16348
"OS friendly name = windows" sip_api.js:1:16348
"Have WebRTC = yes" sip_api.js:1:16348
"Have GUM = yes" sip_api.js:1:16348
"Engine initialized" sip_api.js:1:16348
"s_websocket_server_url=ws://IP:8088/ws" sip_api.js:1:16348
"s_sip_outboundproxy_url=(null)" sip_api.js:1:16348
"b_rtcweb_breaker_enabled=no" sip_api.js:1:16348
"b_click2call_enabled=no" sip_api.js:1:16348
"b_early_ims=no" sip_api.js:1:16348
"b_enable_media_stream_cache=yes" sip_api.js:1:16348
"o_bandwidth={}" sip_api.js:1:16348
"o_video_size={}" sip_api.js:1:16348
"SIP stack start: proxy='ns313841.ovh.net:10060', realm='<sip:IP>', impi='000001', impu='"000001"<sip:000001@IP:8088>'" sip_api.js:1:16348
"Connecting to 'ws://IP:8088/ws'" sip_api.js:1:16348
"|> SipStack iniciado!" IP:3001:765:3
"__tsip_transport_ws_onopen" sip_api.js:1:16348
{id_interno: "1", id_usuario_d: 2670, nome: "Coisa", sexo: "M", foto: "Coisa20150317152721.jpg", cidade: "Santa Cruz do Sul", favorito: true, red_nome: "Bapo", red_icone: "i_logo_bapo.png"} IP:3001:415
"State machine: tsip_dialog_register_Started_2_InProgress_X_oRegister" sip_api.js:1:16348
"SEND: REGISTER sip:IP SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK2zpqnfk8SZSCtFZJhDfT3xeMOrJhkTuX;rport
From: "000001"<sip:000001@IP:8088>;tag=21sD3hs1CpOqrOBO9NsX
To: "000001"<sip:000001@IP:8088>
Contact: "000001"<sip:000001@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=1800;click2call=no
Call-ID: 812515ca-07e6-98e7-9892-f18a83d2aa30
CSeq: 22917 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="000001",realm="IP",nonce="",uri="sip:IP",response=""
User-Agent: IM-client/OMA1.0 sipML5-v1.0.0.0
Organization: Tra La Lala Corp.
Supported: path
" sip_api.js:1:16348
"__tsip_transport_ws_onmessage" sip_api.js:1:16348
"recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=IP; branch=z9hG4bK2zpqnfk8SZSCtFZJhDfT3xeMOrJhkTuX
From: "000001"<sip:000001@IP:8088>;tag=21sD3hs1CpOqrOBO9NsX
To: "000001"<sip:000001@IP:8088>;tag=as427ae015
Call-ID: 812515ca-07e6-98e7-9892-f18a83d2aa30
CSeq: 22917 REGISTER
Content-Length: 0
Server: Asterisk PBX 11.16.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
WWW-Authenticate: Digest realm="IP",nonce="7931c3b8",stale=FALSE,algorithm=MD5
" sip_api.js:1:16348
"State machine: tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494" sip_api.js:1:16348
"SEND: REGISTER sip:IP SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKEXbpaIGIxGtPKm1WPqUtFLnqSlfCNRZB;rport
From: "000001"<sip:000001@IP:8088>;tag=21sD3hs1CpOqrOBO9NsX
To: "000001"<sip:000001@IP:8088>
Contact: "000001"<sip:000001@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=1800;click2call=no
Call-ID: 812515ca-07e6-98e7-9892-f18a83d2aa30
CSeq: 22918 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="000001",realm="IP",nonce="7931c3b8",uri="sip:IP",response="2bd91f75bef84bf128996489ba8b743f",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.0.0.0
Organization: Tra La Lala Corp.
Supported: path
" sip_api.js:1:16348
"__tsip_transport_ws_onmessage" sip_api.js:1:16348
"recv=SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=IP; branch=z9hG4bKEXbpaIGIxGtPKm1WPqUtFLnqSlfCNRZB
From: "000001"<sip:000001@IP:8088>;tag=21sD3hs1CpOqrOBO9NsX
To: "000001"<sip:000001@IP:8088>;tag=as427ae015
Contact: <sip:000001@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=1800
Call-ID: 812515ca-07e6-98e7-9892-f18a83d2aa30
CSeq: 22918 REGISTER
Expires: 1800
Content-Length: 0
Server: Asterisk PBX 11.16.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
Date: 20 Mar 2015 19:08:10 GMT;20
" sip_api.js:1:16348
"State machine: tsip_dialog_register_InProgress_2_Connected_X_2xx"`
所以它正在工作，对于可能遇到此问题的任何其他人，sip.conf 文件缺少一行：
[general]
udpbindaddr=0.0.0.0:5060
realm=IP ;replace with your Asterisk server public IP address or host
rtcachefriends=yes
transport=udp,ws,wss
videosupport=no
avpf=yes
icesupport=yes
directmedia=no
allowguest=no
allwaysreject=yes
rtptimeout=30
allow=all
'rtcachefriends=yes' 似乎是实时所必需的。 在我的 mysql table 中，'qualify' 字段默认为 NULL，应该是。
这就是所有人！