Question

我设置用户“100”和“200”，我正在尝试使用项目sipml5。

服务器在这个在线Linphone上记录两个用户，即用户“100”。

用户“200”在sipml5中记录，但是为了进行调用，它显示了以下错误：

`Mon Dec 14 2015 19:18:51 GMT-0200 (BRST) | sip.invitecontext.mediahandler | 4pk6rpr5p1mhfts149uk9t8l09v01c | acquiring local media`

`Mon Dec 14 2015 19:18:51 GMT-0200 (BRST) | sip.invitecontext.mediahandler | 4pk6rpr5p1mhfts149uk9t8l09v01c | acquired local media streams`

`Mon Dec 14 2015 19:18:51 GMT-0200 (BRST) | sip.invitecontext.mediahandler | 4pk6rpr5p1mhfts149uk9t8l09v01c | ICE candidate received: candidate:3879386089 1 udp 2122260223 172.16.1.209 60702 typ host generation 0`

`Mon Dec 14 2015 19:18:51 GMT-0200 (BRST) | sip.invitecontext.mediahandler | 4pk6rpr5p1mhfts149uk9t8l09v01c | ICE candidate received: candidate:3879386089 2 udp 2122260222 172.16.1.209 45410 typ host generation 0`

`Mon Dec 14 2015 19:18:51 GMT-0200 (BRST) | sip.invitecontext.mediahandler | 4pk6rpr5p1mhfts149uk9t8l09v01c | ICE candidate received: candidate:2847491865 1 tcp 1518280447 172.16.1.209 0 typ host tcptype active generation 0`

`Mon Dec 14 2015 19:18:51 GMT-0200 (BRST) | sip.invitecontext.mediahandler | 4pk6rpr5p1mhfts149uk9t8l09v01c | ICE candidate received: candidate:2847491865 2 tcp 1518280446 172.16.1.209 0 typ host tcptype active generation 0`

`Mon Dec 14 2015 19:18:51 GMT-0200 (BRST) | sip.invitecontext.mediahandler | 4pk6rpr5p1mhfts149uk9t8l09v01c | ICE candidate received: candidate:288014909 2 udp 1686052606 189.35.140.74 62060 typ srflx raddr 172.16.1.209 rport 45410 generation 0`

`Mon Dec 14 2015 19:18:51 GMT-0200 (BRST) | sip.invitecontext.mediahandler | 4pk6rpr5p1mhfts149uk9t8l09v01c | ICE candidate received: candidate:288014909 1 udp 1686052607 189.35.140.74 56282 typ srflx raddr 172.16.1.209 rport 60702 generation 0`

`Mon Dec 14 2015 19:18:51 GMT-0200 (BRST) | sip.transport | sending WebSocket message:`

`INVITE sip:100@172.16.1.203 SIP/2.0<br>
Via: SIP/2.0/WS cam6bslroe6a.invalid;branch=z9hG4bK7117302
Max-Forwards: 70
To: <sip:100@172.16.1.203>
From: "200" <sip:200@172.16.1.203>;tag=ur4c0oov9u
Call-ID: 4pk6rpr5p1mhfts149uk
CSeq: 7760 INVITE
Contact: <sip:040c805k@cam6bslroe6a.invalid;transport=ws;ob>
Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY,INVITE
Content-Type: application/sdp
Supported: outbound
User-Agent: SIP.js/0.7.2
Content-Length: 1777
`

`v=0
o=- 6908994037829104915 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS Mdtr3XiGc3ZYftGDwxeYDfspP9DqQwSRd17N
m=audio 56282 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
c=IN IP4 189.35.140.74
a=rtcp:62060 IN IP4 189.35.140.74
a=candidate:3879386089 1 udp 2122260223 172.16.1.209 60702 typ host generation 0
a=candidate:3879386089 2 udp 2122260222 172.16.1.209 45410 typ host generation 0
a=candidate:2847491865 1 tcp 1518280447 172.16.1.209 0 typ host tcptype active generation 0
a=candidate:2847491865 2 tcp 1518280446 172.16.1.209 0 typ host tcptype active generation 0
a=candidate:288014909 2 udp 1686052606 189.35.140.74 62060 typ srflx raddr 172.16.1.209 rport 45410 generation 0
a=candidate:288014909 1 udp 1686052607 189.35.140.74 56282 typ srflx raddr 172.16.1.209 rport 60702 generation 0
a=ice-ufrag:FK6gzTOP0l95UM39
a=ice-pwd:1ehRZElxNwmCq6mQpetD78w+
a=fingerprint:sha-256 2C:93:D5:54:53:96:34:18:DA:A6:3C:1A:59:5A:32:6B:3E:70:79:78:85:AC:AA:26:8F:D6:0B:5E:9C:DB:4A:8C
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:2011742822 cname:/+L6wPj8ErBqYw3L
a=ssrc:2011742822 msid:Mdtr3XiGc3ZYftGDwxeYDfspP9DqQwSRd17N dce80b6d-4fed-4409-af6c-275e470a0d73
a=ssrc:2011742822 mslabel:Mdtr3XiGc3ZYftGDwxeYDfspP9DqQwSRd17N
a=ssrc:2011742822 label:dce80b6d-4fed-4409-af6c-275e470a0d73`


`Mon Dec 14 2015 19:18:51 GMT-0200 (BRST) | sip.transport | received WebSocket text message:`

`SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS cam6bslroe6a.invalid;branch=z9hG4bK7117302;received=172.16.1.209
From: "200" <sip:200@172.16.1.203>;tag=ur4c0oov9u
To: <sip:100@172.16.1.203>;tag=as5f539d65
Call-ID: 4pk6rpr5p1mhfts149uk
CSeq: 7760 INVITE
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="26be674c"
Content-Length: 0`



`Mon Dec 14 2015 19:18:51 GMT-0200 (BRST) | sip.transport | sending WebSocket message:`

`ACK sip:100@172.16.1.203 SIP/2.0
Via: SIP/2.0/WS cam6bslroe6a.invalid;branch=z9hG4bK7117302
To: <sip:100@172.16.1.203>;tag=as5f539d65
From: "200" <sip:200@172.16.1.203>;tag=ur4c0oov9u
Call-ID: 4pk6rpr5p1mhfts149uk
Content-Length: 0
CSeq: 7760 ACK`



`Mon Dec 14 2015 19:18:51 GMT-0200 (BRST) | sip.transport | sending WebSocket message:`

`INVITE sip:100@172.16.1.203 SIP/2.0
Via: SIP/2.0/WS cam6bslroe6a.invalid;branch=z9hG4bK4757152
Max-Forwards: 70
To: <sip:100@172.16.1.203>
From: "200" <sip:200@172.16.1.203>;tag=ur4c0oov9u
Call-ID: 4pk6rpr5p1mhfts149uk
CSeq: 7761 INVITE
Authorization: Digest algorithm=MD5, username="200", realm="asterisk", nonce="26be674c", uri="sip:100@172.16.1.203", response="7c400e75f1605a4893b99438a18b9560"
Contact: <sip:040c805k@cam6bslroe6a.invalid;transport=ws;ob>
Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY,INVITE
Content-Type: application/sdp
Supported: outbound
User-Agent: SIP.js/0.7.2
Content-Length: 1777`

`v=0
o=- 6908994037829104915 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS Mdtr3XiGc3ZYftGDwxeYDfspP9DqQwSRd17N
m=audio 56282 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
c=IN IP4 189.35.140.74
a=rtcp:62060 IN IP4 189.35.140.74
a=candidate:3879386089 1 udp 2122260223 172.16.1.209 60702 typ host generation 0
a=candidate:3879386089 2 udp 2122260222 172.16.1.209 45410 typ host generation 0
a=candidate:2847491865 1 tcp 1518280447 172.16.1.209 0 typ host tcptype active generation 0
a=candidate:2847491865 2 tcp 1518280446 172.16.1.209 0 typ host tcptype active generation 0
a=candidate:288014909 2 udp 1686052606 189.35.140.74 62060 typ srflx raddr 172.16.1.209 rport 45410 generation 0
a=candidate:288014909 1 udp 1686052607 189.35.140.74 56282 typ srflx raddr 172.16.1.209 rport 60702 generation 0
a=ice-ufrag:FK6gzTOP0l95UM39
a=ice-pwd:1ehRZElxNwmCq6mQpetD78w+
a=fingerprint:sha-256 2C:93:D5:54:53:96:34:18:DA:A6:3C:1A:59:5A:32:6B:3E:70:79:78:85:AC:AA:26:8F:D6:0B:5E:9C:DB:4A:8C
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:2011742822 cname:/+L6wPj8ErBqYw3L
a=ssrc:2011742822 msid:Mdtr3XiGc3ZYftGDwxeYDfspP9DqQwSRd17N dce80b6d-4fed-4409-af6c-275e470a0d73
a=ssrc:2011742822 mslabel:Mdtr3XiGc3ZYftGDwxeYDfspP9DqQwSRd17N
a=ssrc:2011742822 label:dce80b6d-4fed-4409-af6c-275e470a0d73`


`Mon Dec 14 2015 19:18:51 GMT-0200 (BRST) | sip.transaction.ict | z9hG4bK7117302 | Timer D expired for INVITE client transaction z9hG4bK7117302
Mon Dec 14 2015 19:18:51 GMT-0200 (BRST) | sip.transport | received WebSocket text message:`

`SIP/2.0 488 Not acceptable here
Via: SIP/2.0/WS cam6bslroe6a.invalid;branch=z9hG4bK4757152;received=172.16.1.209
From: "200" <sip:200@172.16.1.203>;tag=ur4c0oov9u
To: <sip:100@172.16.1.203>;tag=as5f539d65
Call-ID: 4pk6rpr5p1mhfts149uk
CSeq: 7761 INVITE
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0`



`Mon Dec 14 2015 19:18:51 GMT-0200 (BRST) | sip.transport | sending WebSocket message:`

`ACK sip:100@172.16.1.203 SIP/2.0
Via: SIP/2.0/WS cam6bslroe6a.invalid;branch=z9hG4bK4757152
To: <sip:100@172.16.1.203>;tag=as5f539d65
From: "200" <sip:200@172.16.1.203>;tag=ur4c0oov9u
Call-ID: 4pk6rpr5p1mhfts149uk
Content-Length: 0
CSeq: 7761 ACK`



    `Mon Dec 14 2015 19:18:51 GMT-0200 (BRST) | sip.inviteclientcontext | closing INVITE session 4pk6rpr5p1mhfts149uk9t8l09v01c`

    `Mon Dec 14 2015 19:18:51 GMT-0200 (BRST) | sip.invitecontext.mediahandler | 4pk6rpr5p1mhfts149uk9t8l09v01c | closing PeerConnection`

    `Mon Dec 14 2015 19:18:51 GMT-0200 (BRST) | sip.transaction.ict | z9hG4bK4757152 | Timer D expired for INVITE client transaction z9hG4bK4757152`

Answer 1

488应该意味着没有媒体兼容性。

获得488有几个原因：最常见的是没有常见的编解码器。首先检查您的星号是否支持和配置了opus。

与webrtc兼容，问题通常与使用：

有关

rawlist

而不是

$sftp->nlist('/+mode=text,lrecl=80,recfm=fb');

即：webrtc / ietf标准尚未完全定义且仍在移动。

您可以在此处查看可能重复的问题/答案：Freeswitch and webRTC: media rejected with 488

Answer 2

使用SRTP接听电话时出现此问题。带有488的电话会掉线。我可以毫无问题地拨打加密的电话。

错误“SIP / 2.0 488这里不可接受”

2 个答案: