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ASterisk 11 + webrtc - ошибка "603 Failed to get local SDP"

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ASterisk 11 + webrtc - ошибка "603 Failed to get local SDP"

Добавлено: 11 фев 2014, 20:39
pendolf
Всем привет.

Использую Asterisk 11 с svn и sipml5 script. Проверяю следующим образом. Подключаю на сайте https://sipml5.org/ учетную запись oper1
Звоню с мобильного на номер 0600000000, звонок попадает на elcom затем отправляется в очередь support, членом которого является oper1. Вот oper1 сбрасывает звонок с ошибкой:

== Using SIP RTP CoS mark 5
-- Executing [0622198131@elcat:1] Queue("SIP/elcom-0000001f", "support") in new stack
-- Started music on hold, class 'default', on SIP/elcom-0000001f
== Using SIP RTP CoS mark 5
-- SIP/oper1-00000020 is ringing
-- Got SIP response 603 "Failed to get local SDP" back from 182.42.116.39:58283
-- SIP/oper1-00000020 is busy
-- Nobody picked up in 1000 ms

sip.conf

Код: Выделить всё

[general]

realm=voip.local
transport=udp,ws,wss

videosupport=no
icesupport=yes
context=in
allowguest=no
allowtransfer=no
bindport=5060
bindaddr=0.0.0.0
nat=no
maxexpiry=300
minexpiry=300
defaultexpiry=300
alwaysauthreject=yes
language=ru
disallow=all                                                                                                                                                                     
allow=all                                                                 

useragent=AstPbx                                                                                                                                                                 
dtmfmode = rfc2833                                                                                                                                                               

register => 0600000000:passwd@182.42.114.12/0600000000
registertimeout=300                                                                                                                                                              
registerattempts=10                                                                                                                                                              

[elcom]
type=peer
host=182.42.114.12
defaultuser=0600000000
secret=passwd
canreinvite=no
disallow=all
allow=alaw,ulaw
dtmfmode=rfc2833
insecure=port,invite
qualify=yes

[oper1]
defaultuser=oper1
secret=pass
type=peer
host=dynamic
avpf = yes
allow=all
icesupport = yes
videosupport=no
directmedia=no
qualify=yes
trustrpid=yes
sendrpid=no
qualifyfreq=600
transport=udp,ws
encryption=yes
callcounter=yes
Выхлоп sip set debug peer oper1

Код: Выделить всё

  == Using SIP RTP CoS mark 5
    -- Executing [0622198131@elcom:1] Queue("SIP/elcom-0000001d", "support") in new stack
    -- Started music on hold, class 'default', on SIP/elcom-0000001d
  == Using SIP RTP CoS mark 5
Audio is at 10992
Adding codec 100004 (alaw) to SDP
Adding codec 100001 (g723) to SDP
Adding codec 100001 (g723) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100005 (g726aal2) to SDP
Adding codec 100005 (g726aal2) to SDP
Adding codec 100006 (adpcm) to SDP
Adding codec 100006 (adpcm) to SDP
Adding codec 100007 (lpc10) to SDP
Adding codec 100007 (lpc10) to SDP
Adding codec 100008 (g729) to SDP
Adding codec 100008 (g729) to SDP
Adding codec 100009 (speex) to SDP
Adding codec 100009 (speex) to SDP
Adding codec 100010 (ilbc) to SDP
Adding codec 100010 (ilbc) to SDP
Adding codec 100011 (g726) to SDP
Adding codec 100011 (g726) to SDP
Adding codec 100012 (g722) to SDP
Adding codec 100012 (g722) to SDP
Adding codec 100013 (siren7) to SDP
Adding codec 100013 (siren7) to SDP
Adding codec 100014 (siren14) to SDP
Adding codec 100014 (siren14) to SDP
Adding codec 100015 (g719) to SDP
Adding codec 100015 (g719) to SDP
Adding codec 100016 (speex16) to SDP
Adding codec 100016 (speex16) to SDP
Adding codec 100017 (testlaw) to SDP
Adding codec 100017 (testlaw) to SDP
Adding codec 100018 (silk8) to SDP
Adding codec 100018 (silk12) to SDP
Adding codec 100018 (silk16) to SDP
Adding codec 100018 (silk24) to SDP
Adding codec 100018 (silk8) to SDP
Adding codec 100018 (silk12) to SDP
Adding codec 100018 (silk16) to SDP
Adding codec 100018 (silk24) to SDP
Adding codec 100019 (slin) to SDP
Adding codec 100019 (slin) to SDP
Adding codec 100020 (slin12) to SDP
Adding codec 100020 (slin12) to SDP
Adding codec 100021 (slin16) to SDP
Adding codec 100021 (slin16) to SDP
Adding codec 100022 (slin24) to SDP
Adding codec 100022 (slin24) to SDP
Adding codec 100023 (slin32) to SDP
Adding codec 100023 (slin32) to SDP
Adding codec 100024 (slin44) to SDP
Adding codec 100024 (slin44) to SDP
Adding codec 100025 (slin48) to SDP
Adding codec 100025 (slin48) to SDP
Adding codec 100026 (slin96) to SDP
Adding codec 100026 (slin96) to SDP
Adding codec 100027 (slin192) to SDP
Adding codec 100027 (slin192) to SDP
Adding codec 100028 (speex32) to SDP
Adding codec 100028 (speex32) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 182.42.116.39:58283:
INVITE sip:oper1@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws SIP/2.0
Via: SIP/2.0/WS 182.42.122.200:5060;branch=z9hG4bK2c5cd04c
Max-Forwards: 70
From: <sip:0555401115@182.42.122.200>;tag=as6dc8fa02
To: <sip:oper1@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>
Contact: <sip:0555401115@182.42.122.200:5060;transport=WS>
Call-ID: 6a726a4f2a4713380fd43b551f696324@182.42.122.200:5060
CSeq: 102 INVITE
User-Agent: AstPbx
Date: Tue, 11 Feb 2014 16:25:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 2519

v=0
o=root 1912200572 1912200572 IN IP4 182.42.122.200
s=Asterisk PBX SVN-branch-11-r407874
c=IN IP4 182.42.122.200
t=0 0
m=audio 10992 RTP/SAVPF 8 4 4 3 3 0 0 112 112 5 5 7 7 18 18 110 110 97 97 111 111 9 9 102 102 115 115 116 116 117 117 96 100 107 108 96 100 107 108 10 10 118 118 119 119 101
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:3 GSM/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:116 G719/48000
a=fmtp:116 bitrate=64000
a=rtpmap:116 G719/48000
a=fmtp:116 bitrate=64000
a=rtpmap:117 speex/16000
a=rtpmap:117 speex/16000
a=rtpmap:96 SILK/8000
a=fmtp:96 maxaveragebitrate=10000
a=fmtp:96 usedtx=0
a=fmtp:96 useinbandfec=1
a=rtpmap:100 SILK/12000
a=fmtp:100 maxaveragebitrate=12000
a=fmtp:100 usedtx=0
a=fmtp:100 useinbandfec=1
a=rtpmap:107 SILK/16000
a=fmtp:107 maxaveragebitrate=20000
a=fmtp:107 usedtx=0
a=fmtp:107 useinbandfec=1
a=rtpmap:108 SILK/24000
a=fmtp:108 maxaveragebitrate=30000
a=fmtp:108 usedtx=0
a=fmtp:108 useinbandfec=1
a=rtpmap:96 SILK/8000
a=fmtp:96 maxaveragebitrate=10000
a=fmtp:96 usedtx=0
a=fmtp:96 useinbandfec=1
a=rtpmap:100 SILK/12000
a=fmtp:100 maxaveragebitrate=12000
a=fmtp:100 usedtx=0
a=fmtp:100 useinbandfec=1
a=rtpmap:107 SILK/16000
a=fmtp:107 maxaveragebitrate=20000
a=fmtp:107 usedtx=0
a=fmtp:107 useinbandfec=1
a=rtpmap:108 SILK/24000
a=fmtp:108 maxaveragebitrate=30000
a=fmtp:108 usedtx=0
a=fmtp:108 useinbandfec=1
a=rtpmap:10 L16/8000
a=rtpmap:10 L16/8000
a=rtpmap:118 L16/16000
a=rtpmap:118 L16/16000
a=rtpmap:119 speex/32000
a=rtpmap:119 speex/32000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:svmIrOjO4xRbmntOkDBNQ5w3roBbRTCN0pytsUr/

---

<--- SIP read from WS:182.42.116.39:58283 --->
SIP/2.0 100 Trying (sent from the Transaction Layer)
Via: SIP/2.0/WS 182.42.122.200:5060;branch=z9hG4bK2c5cd04c
From: <sip:0555401115@182.42.122.200>;tag=as6dc8fa02
To: <sip:oper1@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>
Call-ID: 6a726a4f2a4713380fd43b551f696324@182.42.122.200:5060
CSeq: 102 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from WS:182.42.116.39:58283 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WS 182.42.122.200:5060;branch=z9hG4bK2c5cd04c
From: <sip:0555401115@182.42.122.200>;tag=as6dc8fa02
To: <sip:oper1@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=EIexvVo5xQWyM2FxZhtu
Contact: <sip:oper1@df7jal23ls0d.invalid;transport=ws>
Call-ID: 6a726a4f2a4713380fd43b551f696324@182.42.122.200:5060
CSeq: 102 INVITE
Content-Length: 0
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE

<------------->
--- (9 headers 0 lines) ---
list_route: hop: <sip:oper1@df7jal23ls0d.invalid;transport=ws>
    -- SIP/oper1-0000001e is ringing

<--- SIP read from WS:182.42.116.39:58283 --->
SIP/2.0 603 Failed to get local SDP
Via: SIP/2.0/WS 182.42.122.200:5060;branch=z9hG4bK2c5cd04c
From: <sip:0555401115@182.42.122.200>;tag=as6dc8fa02
To: <sip:oper1@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=EIexvVo5xQWyM2FxZhtu
Call-ID: 6a726a4f2a4713380fd43b551f696324@182.42.122.200:5060
CSeq: 102 INVITE
Content-Length: 0
Reason: SIP; cause=603; text="Failed to get local SDP"

<------------->
--- (8 headers 0 lines) ---
    -- Got SIP response 603 "Failed to get local SDP" back from 182.42.116.39:58283
set_destination: Parsing <sip:oper1@df7jal23ls0d.invalid;transport=ws> for address/port to send to
set_destination: URI is for WebSocket, we can't set destination
Transmitting (no NAT) to 182.42.116.39:58283:
ACK sip:oper1@df7jal23ls0d.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 182.42.122.200:5060;branch=z9hG4bK2c5cd04c
Max-Forwards: 70
From: <sip:0555401115@182.42.122.200>;tag=as6dc8fa02
To: <sip:oper1@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=EIexvVo5xQWyM2FxZhtu
Contact: <sip:0555401115@182.42.122.200:5060;transport=WS>
Call-ID: 6a726a4f2a4713380fd43b551f696324@182.42.122.200:5060
CSeq: 102 ACK
User-Agent: AstPbx
Content-Length: 0


---
    -- SIP/oper1-0000001e is busy
    -- Nobody picked up in 0 ms

<--- SIP read from WS:182.42.116.39:58283 --->
SIP/2.0 481 Dialog/Transaction Does Not Exist
Via: SIP/2.0/WS 182.42.122.200:5060;branch=z9hG4bK2c5cd04c
From: <sip:0555401115@182.42.122.200>;tag=as6dc8fa02
To: <sip:oper1@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=EIexvVo5xQWyM2FxZhtu
Call-ID: 6a726a4f2a4713380fd43b551f696324@182.42.122.200:5060
CSeq: 102 ACK
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
[Feb 11 22:25:03] WARNING[30925][C-00000016]: chan_sip.c:23976 handle_response: Remote host can't match request ACK to call '6a726a4f2a4713380fd43b551f696324@182.42.122.200:5060'. Giving up.
Really destroying SIP dialog '6a726a4f2a4713380fd43b551f696324@182.42.122.200:5060' Method: INVITE
    -- Stopped music on hold on SIP/elcom-0000001d
  == Spawn extension (elcom, 0622198131, 1) exited non-zero on 'SIP/elcom-0000001d'
Кто нибудь сталкивался с подобным?
Часовой пояс: UTC+03:00
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