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Настройка WebRTC Asterisk 11.6

Новичком считается только что прочитавший «Астериск - будущее телефонии»
http://asterisk.ru/knowledgebase/books
и пытающийся сделать большее

Модераторы: april22, Zavr2008

Ответить
aleksandr_star
Сообщения: 17
Зарегистрирован: 24 фев 2015, 13:35

Настройка WebRTC Asterisk 11.6

Сообщение aleksandr_star »

Здравствуйте. Пытаюсь настроить WebRTC на Asterisk 11.6. Регистрация WebRTC проходит но исходящие и входящие вызовы не работают. Кто сталкивался с данной проблемой и есть возможность помочь в ее решении буду очень благодарен.

sip.conf

Код: Выделить всё

[general]
udpbindaddr=192.168.1.130:5060
realm=192.168.1.130
dtmfmode=rfc2833
transport=udp,ws,wss
jitterbuffer=yes
nat=force_rport,comedia
nat=yes
tlsenable=yes
tlsbindaddr=192.168.1.130
tlscertfile=/etc/asterisk/keys/asterisk.pem
tlscafile=/etc/asterisk/keys/ca.crt
tlscipher=ALL
tlsclientmethod=tlsv1

[221]
type=friend
context=test
host=dynamic
secret=1234
hasiax=no
hassip=yes
icesupport=yes


[222]
type=friend
username=222
context=test
host=dynamic
secret=1234
encryption=yes
avpf=yes
icesupport=yes
directmedia=no
hasiax=no
hassip=yes
videosupport=no
http.conf

Код: Выделить всё

[general]
enabled=yes
bindaddr=192.168.1.130
bindport=8088
enablestatic=yes
rtp.conf

Код: Выделить всё

[general]
rtpstart=10000
rtpend=20000
icesupport=yes
stunaddr=stun.l.google.com:19302
При входящем звонке когда берешь трубку выдает:
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

Код: Выделить всё

<--- SIP read from UDP:192.168.1.130:43907 --->
INVITE sip:222@192.168.1.130:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.130:43907;branch=z9hG4bK-d8754z-13091cd843631c3a-1---d8754z-
Max-Forwards: 70
Contact: <sip:221@192.168.1.130:43907;transport=UDP>
To: <sip:222@192.168.1.130:5060;transport=UDP>
From: <sip:221@192.168.1.130:5060;transport=UDP>;tag=0e6e676d
Call-ID: MzUyMmI1MDlkZjlmZmRiODNkNzc0OTNkZWZlNmViOTg.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Allow-Events: presence, kpml
Content-Length: 241

v=0
o=Z 0 0 IN IP4 192.168.1.130
s=Z
c=IN IP4 192.168.1.130
t=0 0
m=audio 8000 RTP/AVP 3 110 8 0 98 101
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
Sending to 192.168.1.130:43907 (NAT)
Sending to 192.168.1.130:43907 (NAT)
Using INVITE request as basis request - MzUyMmI1MDlkZjlmZmRiODNkNzc0OTNkZWZlNmViOTg.
Found peer '221' for '221' from 192.168.1.130:43907

<--- Reliably Transmitting (NAT) to 192.168.1.130:43907 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.130:43907;branch=z9hG4bK-d8754z-13091cd843631c3a-1---d8754z-;received=192.168.1.130;rport=43907
From: <sip:221@192.168.1.130:5060;transport=UDP>;tag=0e6e676d
To: <sip:222@192.168.1.130:5060;transport=UDP>;tag=as51f771a8
Call-ID: MzUyMmI1MDlkZjlmZmRiODNkNzc0OTNkZWZlNmViOTg.
CSeq: 1 INVITE
Server: Asterisk PBX 11.6-cert10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="192.168.1.130", nonce="065f3a34"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'MzUyMmI1MDlkZjlmZmRiODNkNzc0OTNkZWZlNmViOTg.' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.1.130:43907 --->
PUBLISH sip:221@192.168.1.130:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.130:43907;branch=z9hG4bK-d8754z-40b236a7b882c08e-1---d8754z-
Max-Forwards: 70
Contact: <sip:221@192.168.1.130:43907;transport=UDP>
To: <sip:221@192.168.1.130:5060;transport=UDP>
From: <sip:221@192.168.1.130:5060;transport=UDP>;tag=0653b571
Call-ID: ZDIwNWRiNmMwNjBjYjJiOWVlNDEwZGExMGJiOTZhNDc.
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Event: presence
Allow-Events: presence, kpml
Content-Length: 270

<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf" entity="sip:221@192.168.1.130:5060;transport=UDP"> <tuple id="221" > <status><basic>open</basic></status> <note>On the phone</note> </tuple>
</presence>
<------------->
--- (16 headers 3 lines) ---
Sending to 192.168.1.130:43907 (NAT)

<--- Transmitting (NAT) to 192.168.1.130:43907 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.1.130:43907;branch=z9hG4bK-d8754z-40b236a7b882c08e-1---d8754z-;received=192.168.1.130;rport=43907
From: <sip:221@192.168.1.130:5060;transport=UDP>;tag=0653b571
To: <sip:221@192.168.1.130:5060;transport=UDP>;tag=as5d61e97a
Call-ID: ZDIwNWRiNmMwNjBjYjJiOWVlNDEwZGExMGJiOTZhNDc.
CSeq: 1 PUBLISH
Server: Asterisk PBX 11.6-cert10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog 'ZDIwNWRiNmMwNjBjYjJiOWVlNDEwZGExMGJiOTZhNDc.' Method: PUBLISH

<--- SIP read from UDP:192.168.1.130:43907 --->
SUBSCRIBE sip:221@192.168.1.130:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.130:43907;branch=z9hG4bK-d8754z-e0f0024fc881ca47-1---d8754z-
Max-Forwards: 70
Contact: <sip:221@192.168.1.130:43907;transport=UDP>
To: <sip:221@192.168.1.130:5060;transport=UDP>
From: <sip:221@192.168.1.130:5060;transport=UDP>;tag=2e430e52
Call-ID: N2VhZjczZTEzYjVjY2M4N2ZiNDFhYzIwNDU2YmU1Y2Y.
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (16 headers 0 lines) ---
Sending to 192.168.1.130:43907 (NAT)
Creating new subscription
Sending to 192.168.1.130:43907 (NAT)
list_route: hop: <sip:221@192.168.1.130:43907;transport=UDP>
Found peer '221' for '221' from 192.168.1.130:43907

<--- Transmitting (NAT) to 192.168.1.130:43907 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.130:43907;branch=z9hG4bK-d8754z-e0f0024fc881ca47-1---d8754z-;received=192.168.1.130;rport=43907
From: <sip:221@192.168.1.130:5060;transport=UDP>;tag=2e430e52
To: <sip:221@192.168.1.130:5060;transport=UDP>;tag=as5512ff7b
Call-ID: N2VhZjczZTEzYjVjY2M4N2ZiNDFhYzIwNDU2YmU1Y2Y.
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX 11.6-cert10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="192.168.1.130", nonce="44be56ed"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'N2VhZjczZTEzYjVjY2M4N2ZiNDFhYzIwNDU2YmU1Y2Y.' in 32000 ms (Method: SUBSCRIBE)

<--- SIP read from UDP:192.168.1.130:43907 --->
ACK sip:222@192.168.1.130:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.130:43907;branch=z9hG4bK-d8754z-13091cd843631c3a-1---d8754z-
Max-Forwards: 70
To: <sip:222@192.168.1.130:5060;transport=UDP>;tag=as51f771a8
From: <sip:221@192.168.1.130:5060;transport=UDP>;tag=0e6e676d
Call-ID: MzUyMmI1MDlkZjlmZmRiODNkNzc0OTNkZWZlNmViOTg.
CSeq: 1 ACK
Content-Length: 0

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

<--- SIP read from UDP:192.168.1.130:43907 --->
INVITE sip:222@192.168.1.130:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.130:43907;branch=z9hG4bK-d8754z-76c42c5a54330359-1---d8754z-
Max-Forwards: 70
Contact: <sip:221@192.168.1.130:43907;transport=UDP>
To: <sip:222@192.168.1.130:5060;transport=UDP>
From: <sip:221@192.168.1.130:5060;transport=UDP>;tag=0e6e676d
Call-ID: MzUyMmI1MDlkZjlmZmRiODNkNzc0OTNkZWZlNmViOTg.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Authorization: Digest username="221",realm="192.168.1.130",nonce="065f3a34",uri="sip:222@192.168.1.130:5060;transport=UDP",response="9a290dc60ad2765a72711aa78b2c0190",algorithm=MD5
Allow-Events: presence, kpml
Content-Length: 241

v=0
o=Z 0 0 IN IP4 192.168.1.130
s=Z
c=IN IP4 192.168.1.130
t=0 0
m=audio 8000 RTP/AVP 3 110 8 0 98 101
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (15 headers 12 lines) ---
Sending to 192.168.1.130:43907 (NAT)
Using INVITE request as basis request - MzUyMmI1MDlkZjlmZmRiODNkNzc0OTNkZWZlNmViOTg.
Found peer '221' for '221' from 192.168.1.130:43907
  == Using SIP RTP CoS mark 5
Found RTP audio format 3
Found RTP audio format 110
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 98
Found RTP audio format 101
Found audio description format speex for ID 110
Found audio description format iLBC for ID 98
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(gsm|ulaw|alaw|speex|ilbc)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.130:8000
Looking for 222 in test (domain 192.168.1.130)
list_route: hop: <sip:221@192.168.1.130:43907;transport=UDP>

<--- Transmitting (NAT) to 192.168.1.130:43907 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.130:43907;branch=z9hG4bK-d8754z-76c42c5a54330359-1---d8754z-;received=192.168.1.130;rport=43907
From: <sip:221@192.168.1.130:5060;transport=UDP>;tag=0e6e676d
To: <sip:222@192.168.1.130:5060;transport=UDP>
Call-ID: MzUyMmI1MDlkZjlmZmRiODNkNzc0OTNkZWZlNmViOTg.
CSeq: 2 INVITE
Server: Asterisk PBX 11.6-cert10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:222@192.168.1.130:5060>
Content-Length: 0


<------------>

<--- SIP read from UDP:192.168.1.130:43907 --->
SUBSCRIBE sip:221@192.168.1.130:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.130:43907;branch=z9hG4bK-d8754z-14692aca0c10c4a2-1---d8754z-
Max-Forwards: 70
Contact: <sip:221@192.168.1.130:43907;transport=UDP>
To: <sip:221@192.168.1.130:5060;transport=UDP>
From: <sip:221@192.168.1.130:5060;transport=UDP>;tag=2e430e52
Call-ID: N2VhZjczZTEzYjVjY2M4N2ZiNDFhYzIwNDU2YmU1Y2Y.
CSeq: 2 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Authorization: Digest username="221",realm="192.168.1.130",nonce="44be56ed",uri="sip:221@192.168.1.130:5060;transport=UDP",response="5f17a3734768205cd102928a228c8ecb",algorithm=MD5
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (17 headers 0 lines) ---
Creating new subscription
Sending to 192.168.1.130:43907 (NAT)
Found peer '221' for '221' from 192.168.1.130:43907
    -- Executing [222@test:1] Verbose("SIP/221-00000004", "") in new stack
    -- Executing [222@test:2] Dial("SIP/221-00000004", "SIP/222") in new stack

<--- Transmitting (NAT) to 192.168.1.130:43907 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.1.130:43907;branch=z9hG4bK-d8754z-14692aca0c10c4a2-1---d8754z-;received=192.168.1.130;rport=43907
From: <sip:221@192.168.1.130:5060;transport=UDP>;tag=2e430e52
To: <sip:221@192.168.1.130:5060;transport=UDP>;tag=as5512ff7b
Call-ID: N2VhZjczZTEzYjVjY2M4N2ZiNDFhYzIwNDU2YmU1Y2Y.
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX 11.6-cert10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog 'N2VhZjczZTEzYjVjY2M4N2ZiNDFhYzIwNDU2YmU1Y2Y.' Method: SUBSCRIBE
  == Using SIP RTP CoS mark 5
Audio is at 19290
Adding codec 100003 (ulaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100017 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.1.130:58842:
INVITE sip:222@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.1.130:5060;branch=z9hG4bK69f19251;rport
Max-Forwards: 70
From: <sip:221@192.168.1.130>;tag=as4eaceecf
To: <sip:222@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>
Contact: <sip:221@192.168.1.130:5060;transport=WS>
Call-ID: 3a7532c86272ba1f10ff06b807d26bf8@192.168.1.130:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.6-cert10
Date: Tue, 24 Feb 2015 19:10:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 768

v=0
o=root 1690592487 1690592487 IN IP4 192.168.1.130
s=Asterisk PBX 11.6-cert10
c=IN IP4 192.168.1.130
t=0 0
m=audio 19290 RTP/SAVPF 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=ice-ufrag:3de81acb14829ba457cae93b2303c8a7
a=ice-pwd:1fa81ff73056812f38ae27bd6cab6767
a=candidate:Hc0a80182 1 UDP 2130706431 192.168.1.130 19290 typ host
a=candidate:S4d5729c6 1 UDP 1694498815 77.87.41.198 19290 typ srflx
a=candidate:Hc0a80182 2 UDP 2130706430 192.168.1.130 19291 typ host
a=candidate:S4d5729c6 2 UDP 1694498814 77.87.41.198 19292 typ srflx
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:vy4ITRW/ax4RBOfI0a5DPzF/AmUEtprUhGX30PR7

---
    -- Called SIP/222

<--- SIP read from WS:192.168.1.130:58842 --->
SIP/2.0 100 Trying (sent from the Transaction Layer)
Via: SIP/2.0/WS 192.168.1.130:5060;rport=5060;branch=z9hG4bK69f19251
From: <sip:221@192.168.1.130>;tag=as4eaceecf
To: <sip:222@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>
Call-ID: 3a7532c86272ba1f10ff06b807d26bf8@192.168.1.130:5060
CSeq: 102 INVITE
Content-Length: 0

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

<--- SIP read from WS:192.168.1.130:58842 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WS 192.168.1.130:5060;rport=5060;branch=z9hG4bK69f19251
From: <sip:221@192.168.1.130>;tag=as4eaceecf
To: <sip:222@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=GnvnZIGce113IrRflZ8l
Contact: <sip:222@df7jal23ls0d.invalid;transport=ws>
Call-ID: 3a7532c86272ba1f10ff06b807d26bf8@192.168.1.130:5060
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:222@df7jal23ls0d.invalid;transport=ws>
    -- SIP/222-00000005 is ringing

<--- Transmitting (NAT) to 192.168.1.130:43907 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.130:43907;branch=z9hG4bK-d8754z-76c42c5a54330359-1---d8754z-;received=192.168.1.130;rport=43907
From: <sip:221@192.168.1.130:5060;transport=UDP>;tag=0e6e676d
To: <sip:222@192.168.1.130:5060;transport=UDP>;tag=as5a29d8af
Call-ID: MzUyMmI1MDlkZjlmZmRiODNkNzc0OTNkZWZlNmViOTg.
CSeq: 2 INVITE
Server: Asterisk PBX 11.6-cert10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:222@192.168.1.130:5060>
Content-Length: 0


<------------>

<--- SIP read from WS:192.168.1.130:58842 --->
SIP/2.0 603 Failed to get local SDP
Via: SIP/2.0/WS 192.168.1.130:5060;rport=5060;branch=z9hG4bK69f19251
From: <sip:221@192.168.1.130>;tag=as4eaceecf
To: <sip:222@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=GnvnZIGce113IrRflZ8l
Call-ID: 3a7532c86272ba1f10ff06b807d26bf8@192.168.1.130:5060
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 192.168.1.130:58842
set_destination: Parsing <sip:222@df7jal23ls0d.invalid;transport=ws> for address/port to send to
set_destination: URI is for WebSocket, we can't set destination
Transmitting (NAT) to 192.168.1.130:58842:
ACK sip:222@df7jal23ls0d.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.1.130:5060;branch=z9hG4bK69f19251;rport
Max-Forwards: 70
From: <sip:221@192.168.1.130>;tag=as4eaceecf
To: <sip:222@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=GnvnZIGce113IrRflZ8l
Contact: <sip:221@192.168.1.130:5060;transport=WS>
Call-ID: 3a7532c86272ba1f10ff06b807d26bf8@192.168.1.130:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.6-cert10
Content-Length: 0


---
    -- SIP/222-00000005 is busy
  == Everyone is busy/congested at this time (1:1/0/0)
    -- Executing [222@test:3] Hangup("SIP/221-00000004", "") in new stack
  == Spawn extension (test, 222, 3) exited non-zero on 'SIP/221-00000004'
Scheduling destruction of SIP dialog 'MzUyMmI1MDlkZjlmZmRiODNkNzc0OTNkZWZlNmViOTg.' in 32000 ms (Method: INVITE)

<--- Reliably Transmitting (NAT) to 192.168.1.130:43907 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.1.130:43907;branch=z9hG4bK-d8754z-76c42c5a54330359-1---d8754z-;received=192.168.1.130;rport=43907
From: <sip:221@192.168.1.130:5060;transport=UDP>;tag=0e6e676d
To: <sip:222@192.168.1.130:5060;transport=UDP>;tag=as5a29d8af
Call-ID: MzUyMmI1MDlkZjlmZmRiODNkNzc0OTNkZWZlNmViOTg.
CSeq: 2 INVITE
Server: Asterisk PBX 11.6-cert10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog '3a7532c86272ba1f10ff06b807d26bf8@192.168.1.130:5060' Method: INVITE

<--- SIP read from UDP:192.168.1.130:43907 --->
ACK sip:222@192.168.1.130:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.130:43907;branch=z9hG4bK-d8754z-76c42c5a54330359-1---d8754z-
Max-Forwards: 70
To: <sip:222@192.168.1.130:5060;transport=UDP>;tag=as5a29d8af
From: <sip:221@192.168.1.130:5060;transport=UDP>;tag=0e6e676d
Call-ID: MzUyMmI1MDlkZjlmZmRiODNkNzc0OTNkZWZlNmViOTg.
CSeq: 2 ACK
Content-Length: 0

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

<--- SIP read from WS:192.168.1.130:58842 --->
SIP/2.0 481 Dialog/Transaction Does Not Exist
Via: SIP/2.0/WS 192.168.1.130:5060;rport=5060;branch=z9hG4bK69f19251
From: <sip:221@192.168.1.130>;tag=as4eaceecf
To: <sip:222@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=GnvnZIGce113IrRflZ8l
Call-ID: 3a7532c86272ba1f10ff06b807d26bf8@192.168.1.130:5060
CSeq: 102 ACK
Content-Length: 0

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

<--- SIP read from UDP:192.168.1.130:43907 --->
PUBLISH sip:221@192.168.1.130:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.130:43907;branch=z9hG4bK-d8754z-04c45d1ab425030e-1---d8754z-
Max-Forwards: 70
Contact: <sip:221@192.168.1.130:43907;transport=UDP>
To: <sip:221@192.168.1.130:5060;transport=UDP>
From: <sip:221@192.168.1.130:5060;transport=UDP>;tag=f6d61549
Call-ID: MThmN2UyNWM0N2Y4N2NkNTE4NTBmNzExNmI2YmY5MWQ.
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Event: presence
Allow-Events: presence, kpml
Content-Length: 264

<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf" entity="sip:221@192.168.1.130:5060;transport=UDP"> <tuple id="221" > <status><basic>open</basic></status> <note>Online</note> </tuple>
</presence>
<------------->
--- (16 headers 3 lines) ---
Sending to 192.168.1.130:43907 (NAT)

<--- Transmitting (NAT) to 192.168.1.130:43907 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.1.130:43907;branch=z9hG4bK-d8754z-04c45d1ab425030e-1---d8754z-;received=192.168.1.130;rport=43907
From: <sip:221@192.168.1.130:5060;transport=UDP>;tag=f6d61549
To: <sip:221@192.168.1.130:5060;transport=UDP>;tag=as565bbe52
Call-ID: MThmN2UyNWM0N2Y4N2NkNTE4NTBmNzExNmI2YmY5MWQ.
CSeq: 1 PUBLISH
Server: Asterisk PBX 11.6-cert10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog 'MThmN2UyNWM0N2Y4N2NkNTE4NTBmNzExNmI2YmY5MWQ.' Method: PUBLISH

<--- SIP read from UDP:192.168.1.130:43907 --->
SUBSCRIBE sip:221@192.168.1.130:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.130:43907;branch=z9hG4bK-d8754z-ca9b84d6af0f69cf-1---d8754z-
Max-Forwards: 70
Contact: <sip:221@192.168.1.130:43907;transport=UDP>
To: <sip:221@192.168.1.130:5060;transport=UDP>
From: <sip:221@192.168.1.130:5060;transport=UDP>;tag=75b02121
Call-ID: Mzc4YzI0MmYyODY0NWQ1NGVkZTFhNGU4MGIzYjdmMzQ.
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (16 headers 0 lines) ---
Sending to 192.168.1.130:43907 (NAT)
Creating new subscription
Sending to 192.168.1.130:43907 (NAT)
list_route: hop: <sip:221@192.168.1.130:43907;transport=UDP>
Found peer '221' for '221' from 192.168.1.130:43907

<--- Transmitting (NAT) to 192.168.1.130:43907 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.130:43907;branch=z9hG4bK-d8754z-ca9b84d6af0f69cf-1---d8754z-;received=192.168.1.130;rport=43907
From: <sip:221@192.168.1.130:5060;transport=UDP>;tag=75b02121
To: <sip:221@192.168.1.130:5060;transport=UDP>;tag=as59b61d54
Call-ID: Mzc4YzI0MmYyODY0NWQ1NGVkZTFhNGU4MGIzYjdmMzQ.
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX 11.6-cert10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="192.168.1.130", nonce="6a1d6abe"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'Mzc4YzI0MmYyODY0NWQ1NGVkZTFhNGU4MGIzYjdmMzQ.' in 32000 ms (Method: SUBSCRIBE)

<--- SIP read from UDP:192.168.1.130:43907 --->
SUBSCRIBE sip:221@192.168.1.130:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.130:43907;branch=z9hG4bK-d8754z-d211c5c603479dbb-1---d8754z-
Max-Forwards: 70
Contact: <sip:221@192.168.1.130:43907;transport=UDP>
To: <sip:221@192.168.1.130:5060;transport=UDP>
From: <sip:221@192.168.1.130:5060;transport=UDP>;tag=75b02121
Call-ID: Mzc4YzI0MmYyODY0NWQ1NGVkZTFhNGU4MGIzYjdmMzQ.
CSeq: 2 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Authorization: Digest username="221",realm="192.168.1.130",nonce="6a1d6abe",uri="sip:221@192.168.1.130:5060;transport=UDP",response="a21e4b3d397226cdac3601ad3c45a06a",algorithm=MD5
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (17 headers 0 lines) ---
Creating new subscription
Sending to 192.168.1.130:43907 (NAT)
Found peer '221' for '221' from 192.168.1.130:43907

<--- Transmitting (NAT) to 192.168.1.130:43907 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.1.130:43907;branch=z9hG4bK-d8754z-d211c5c603479dbb-1---d8754z-;received=192.168.1.130;rport=43907
From: <sip:221@192.168.1.130:5060;transport=UDP>;tag=75b02121
To: <sip:221@192.168.1.130:5060;transport=UDP>;tag=as59b61d54
Call-ID: Mzc4YzI0MmYyODY0NWQ1NGVkZTFhNGU4MGIzYjdmMzQ.
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX 11.6-cert10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
А при исходящем:
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

Код: Выделить всё

<--- SIP read from WS:192.168.1.130:58923 --->
INVITE sip:223@192.168.1.130 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKAsUzdFau40GoFuiVpdDyPwyxDSIF3CLq;rport
From: "222"<sip:222@192.168.1.130:5060>;tag=3QqgtL5UqsDQGCHSvp0G
To: <sip:223@192.168.1.130>
Contact: "222"<sip:222@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;impi=222;ha1=4721514ea9a1ed1efe8524e5b40c7a57;+g.oma.sip-im;language="en,fr"
Call-ID: 1df90949-77a2-cca8-3410-f23a44fb0bdb
CSeq: 19146 INVITE
Content-Type: application/sdp
Content-Length: 1815
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
Organization: Doubango Telecom

v=0
o=- 5726801840161240000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS xyGNsvhBRRdngtpGATI7xZjsKW9DYoQJwjTP
m=audio 33600 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
c=IN IP4 77.87.41.198
a=rtcp:33600 IN IP4 77.87.41.198
a=candidate:4093981320 1 udp 2122260223 192.168.1.130 33600 typ host generation 0
a=candidate:4093981320 2 udp 2122260223 192.168.1.130 33600 typ host generation 0
a=candidate:1967993916 1 udp 1686052607 77.87.41.198 33600 typ srflx raddr 192.168.1.130 rport 33600 generation 0
a=candidate:1967993916 2 udp 1686052607 77.87.41.198 33600 typ srflx raddr 192.168.1.130 rport 33600 generation 0
a=candidate:3129396856 1 tcp 1518280447 192.168.1.130 0 typ host tcptype active generation 0
a=candidate:3129396856 2 tcp 1518280447 192.168.1.130 0 typ host tcptype active generation 0
a=ice-ufrag:bhGDqN4uEIwMbABK
a=ice-pwd:14C1kJaoaL039LPvLWmKdmok
a=ice-options:google-ice
a=fingerprint:sha-256 29:42:5A:38:C0:3B:8C:D6:F7:60:4F:42:92:E5:7A:51:A3:0C:AB:82:DF:7B:0F:1B:98:E2:2B:B5:D7:C2:0A:84
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:1533397640 cname:OGE6DiPDm73cQU9Z
a=ssrc:1533397640 msid:xyGNsvhBRRdngtpGATI7xZjsKW9DYoQJwjTP 2ab36d79-5e20-46f0-ba38-89d202793aa4
a=ssrc:1533397640 mslabel:xyGNsvhBRRdngtpGATI7xZjsKW9DYoQJwjTP
a=ssrc:1533397640 label:2ab36d79-5e20-46f0-ba38-89d202793aa4
<------------->
--- (12 headers 41 lines) ---
Using INVITE request as basis request - 1df90949-77a2-cca8-3410-f23a44fb0bdb
Found peer '222' for '222' from 192.168.1.130:58923

<--- Reliably Transmitting (NAT) to 192.168.1.130:58923 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKAsUzdFau40GoFuiVpdDyPwyxDSIF3CLq;received=192.168.1.130;rport=58923
From: "222"<sip:222@192.168.1.130:5060>;tag=3QqgtL5UqsDQGCHSvp0G
To: <sip:223@192.168.1.130>;tag=as513ab50f
Call-ID: 1df90949-77a2-cca8-3410-f23a44fb0bdb
CSeq: 19146 INVITE
Server: Asterisk PBX 11.6-cert10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="192.168.1.130", nonce="5273e20c"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1df90949-77a2-cca8-3410-f23a44fb0bdb' in 32000 ms (Method: INVITE)

<--- SIP read from WS:192.168.1.130:58923 --->
ACK sip:223@192.168.1.130 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKAsUzdFau40GoFuiVpdDyPwyxDSIF3CLq;rport
From: "222"<sip:222@192.168.1.130:5060>;tag=3QqgtL5UqsDQGCHSvp0G
To: <sip:223@192.168.1.130>;tag=as513ab50f
Call-ID: 1df90949-77a2-cca8-3410-f23a44fb0bdb
CSeq: 19146 ACK
Content-Length: 0
Max-Forwards: 70

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

<--- SIP read from WS:192.168.1.130:58923 --->
INVITE sip:223@192.168.1.130 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKEErNcbxay2qom02fIQCtXA7H3HPU2MtP;rport
From: "222"<sip:222@192.168.1.130:5060>;tag=3QqgtL5UqsDQGCHSvp0G
To: <sip:223@192.168.1.130>
Contact: "222"<sip:222@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;impi=222;ha1=4721514ea9a1ed1efe8524e5b40c7a57;+g.oma.sip-im;language="en,fr"
Call-ID: 1df90949-77a2-cca8-3410-f23a44fb0bdb
CSeq: 19147 INVITE
Content-Type: application/sdp
Content-Length: 1815
Max-Forwards: 70
Authorization: Digest username="222",realm="192.168.1.130",nonce="5273e20c",uri="sip:223@192.168.1.130",response="32f3534284ab7a508b22fe05d3d16a37",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
Organization: Doubango Telecom

v=0
o=- 5726801840161240000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS xyGNsvhBRRdngtpGATI7xZjsKW9DYoQJwjTP
m=audio 33600 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
c=IN IP4 77.87.41.198
a=rtcp:33600 IN IP4 77.87.41.198
a=candidate:4093981320 1 udp 2122260223 192.168.1.130 33600 typ host generation 0
a=candidate:4093981320 2 udp 2122260223 192.168.1.130 33600 typ host generation 0
a=candidate:1967993916 1 udp 1686052607 77.87.41.198 33600 typ srflx raddr 192.168.1.130 rport 33600 generation 0
a=candidate:1967993916 2 udp 1686052607 77.87.41.198 33600 typ srflx raddr 192.168.1.130 rport 33600 generation 0
a=candidate:3129396856 1 tcp 1518280447 192.168.1.130 0 typ host tcptype active generation 0
a=candidate:3129396856 2 tcp 1518280447 192.168.1.130 0 typ host tcptype active generation 0
a=ice-ufrag:bhGDqN4uEIwMbABK
a=ice-pwd:14C1kJaoaL039LPvLWmKdmok
a=ice-options:google-ice
a=fingerprint:sha-256 29:42:5A:38:C0:3B:8C:D6:F7:60:4F:42:92:E5:7A:51:A3:0C:AB:82:DF:7B:0F:1B:98:E2:2B:B5:D7:C2:0A:84
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:1533397640 cname:OGE6DiPDm73cQU9Z
a=ssrc:1533397640 msid:xyGNsvhBRRdngtpGATI7xZjsKW9DYoQJwjTP 2ab36d79-5e20-46f0-ba38-89d202793aa4
a=ssrc:1533397640 mslabel:xyGNsvhBRRdngtpGATI7xZjsKW9DYoQJwjTP
a=ssrc:1533397640 label:2ab36d79-5e20-46f0-ba38-89d202793aa4
<------------->
--- (13 headers 41 lines) ---
Using INVITE request as basis request - 1df90949-77a2-cca8-3410-f23a44fb0bdb
Found peer '222' for '222' from 192.168.1.130:58923
  == Using SIP RTP CoS mark 5
Found RTP audio format 111
Found RTP audio format 103
Found RTP audio format 104
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 106
Found RTP audio format 105
Found RTP audio format 13
Found RTP audio format 126
Found unknown media description format opus for ID 111
Found unknown media description format ISAC for ID 103
Found unknown media description format ISAC for ID 104
Found audio description format G722 for ID 9
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found unknown media description format CN for ID 106
Found unknown media description format CN for ID 105
Found audio description format CN for ID 13
Found audio description format telephone-event for ID 126

<--- Reliably Transmitting (NAT) to 192.168.1.130:58923 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKEErNcbxay2qom02fIQCtXA7H3HPU2MtP;received=192.168.1.130;rport=58923
From: "222"<sip:222@192.168.1.130:5060>;tag=3QqgtL5UqsDQGCHSvp0G
To: <sip:223@192.168.1.130>;tag=as513ab50f
Call-ID: 1df90949-77a2-cca8-3410-f23a44fb0bdb
CSeq: 19147 INVITE
Server: Asterisk PBX 11.6-cert10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1df90949-77a2-cca8-3410-f23a44fb0bdb' in 32000 ms (Method: INVITE)

<--- SIP read from WS:192.168.1.130:58923 --->
ACK sip:223@192.168.1.130 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKEErNcbxay2qom02fIQCtXA7H3HPU2MtP;rport
From: "222"<sip:222@192.168.1.130:5060>;tag=3QqgtL5UqsDQGCHSvp0G
To: <sip:223@192.168.1.130>;tag=as513ab50f
Call-ID: 1df90949-77a2-cca8-3410-f23a44fb0bdb
CSeq: 19147 ACK
Content-Length: 0
Max-Forwards: 70

<------------->
--- (8 headers 0 lines) ---
Aleksandr-K53SD*CLI> 
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups
aleksandr@Aleksandr-K53SD:/etc/asterisk$ asterisk -rvvvv
Asterisk 11.6-cert10, Copyright (C) 1999 - 2013 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Running as user 'aleksandr'
Running under group 'aleksandr'
Connected to Asterisk 11.6-cert10 currently running on Aleksandr-K53SD (pid = 18890)
Aleksandr-K53SD*CLI> 
Aleksandr-K53SD*CLI> 
Aleksandr-K53SD*CLI> 
Aleksandr-K53SD*CLI> 
Aleksandr-K53SD*CLI> 
Aleksandr-K53SD*CLI> 
Aleksandr-K53SD*CLI> 
Aleksandr-K53SD*CLI> 
Aleksandr-K53SD*CLI> 
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Really destroying SIP dialog 'MjA2NDFiMWU1Njg3NDJjOWI0OTViMjgzY2UyNGU3MjI.' Method: REGISTER
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Aleksandr-K53SD*CLI> 

<--- SIP read from WS:192.168.1.130:58923 --->
INVITE sip:221@192.168.1.130 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKzXTwcfluX74JTFJPPX1raX8ao3q119fP;rport
From: "222"<sip:222@192.168.1.130:5060>;tag=54hy0vZSvTcIz2AUWGxC
To: <sip:221@192.168.1.130>
Contact: "222"<sip:222@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;impi=222;ha1=4721514ea9a1ed1efe8524e5b40c7a57;+g.oma.sip-im;language="en,fr"
Call-ID: ff63e03b-6e45-0131-025f-4acae05c063e
CSeq: 8121 INVITE
Content-Type: application/sdp
Content-Length: 1815
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
Organization: Doubango Telecom

v=0
o=- 1708632238310997200 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS RjZC3RuRaCy7ZEn7QF97pDs2ZARIy9yBP6Sy
m=audio 56550 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
c=IN IP4 77.87.41.198
a=rtcp:56550 IN IP4 77.87.41.198
a=candidate:4093981320 1 udp 2122260223 192.168.1.130 56550 typ host generation 0
a=candidate:4093981320 2 udp 2122260223 192.168.1.130 56550 typ host generation 0
a=candidate:1967993916 1 udp 1686052607 77.87.41.198 56550 typ srflx raddr 192.168.1.130 rport 56550 generation 0
a=candidate:1967993916 2 udp 1686052607 77.87.41.198 56550 typ srflx raddr 192.168.1.130 rport 56550 generation 0
a=candidate:3129396856 1 tcp 1518280447 192.168.1.130 0 typ host tcptype active generation 0
a=candidate:3129396856 2 tcp 1518280447 192.168.1.130 0 typ host tcptype active generation 0
a=ice-ufrag:S/vjEA1xTE4ilg1Z
a=ice-pwd:P3jwK0NS4JpZ3pe3LF3l+O4j
a=ice-options:google-ice
a=fingerprint:sha-256 29:42:5A:38:C0:3B:8C:D6:F7:60:4F:42:92:E5:7A:51:A3:0C:AB:82:DF:7B:0F:1B:98:E2:2B:B5:D7:C2:0A:84
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:2020025942 cname:F2asousgHTFy3qBT
a=ssrc:2020025942 msid:RjZC3RuRaCy7ZEn7QF97pDs2ZARIy9yBP6Sy 8071dd9b-8fb0-43a5-8969-daac77bf06c2
a=ssrc:2020025942 mslabel:RjZC3RuRaCy7ZEn7QF97pDs2ZARIy9yBP6Sy
a=ssrc:2020025942 label:8071dd9b-8fb0-43a5-8969-daac77bf06c2
<------------->
--- (12 headers 41 lines) ---
Using INVITE request as basis request - ff63e03b-6e45-0131-025f-4acae05c063e
Found peer '222' for '222' from 192.168.1.130:58923

<--- Reliably Transmitting (NAT) to 192.168.1.130:58923 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKzXTwcfluX74JTFJPPX1raX8ao3q119fP;received=192.168.1.130;rport=58923
From: "222"<sip:222@192.168.1.130:5060>;tag=54hy0vZSvTcIz2AUWGxC
To: <sip:221@192.168.1.130>;tag=as5e7d639f
Call-ID: ff63e03b-6e45-0131-025f-4acae05c063e
CSeq: 8121 INVITE
Server: Asterisk PBX 11.6-cert10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="192.168.1.130", nonce="1489bb04"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'ff63e03b-6e45-0131-025f-4acae05c063e' in 32000 ms (Method: INVITE)

<--- SIP read from WS:192.168.1.130:58923 --->
ACK sip:221@192.168.1.130 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKzXTwcfluX74JTFJPPX1raX8ao3q119fP;rport
From: "222"<sip:222@192.168.1.130:5060>;tag=54hy0vZSvTcIz2AUWGxC
To: <sip:221@192.168.1.130>;tag=as5e7d639f
Call-ID: ff63e03b-6e45-0131-025f-4acae05c063e
CSeq: 8121 ACK
Content-Length: 0
Max-Forwards: 70

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

<--- SIP read from WS:192.168.1.130:58923 --->
INVITE sip:221@192.168.1.130 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKFBebVekgl49yqdLXslwnmxS6dPccTxft;rport
From: "222"<sip:222@192.168.1.130:5060>;tag=54hy0vZSvTcIz2AUWGxC
To: <sip:221@192.168.1.130>
Contact: "222"<sip:222@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;impi=222;ha1=4721514ea9a1ed1efe8524e5b40c7a57;+g.oma.sip-im;language="en,fr"
Call-ID: ff63e03b-6e45-0131-025f-4acae05c063e
CSeq: 8122 INVITE
Content-Type: application/sdp
Content-Length: 1815
Max-Forwards: 70
Authorization: Digest username="222",realm="192.168.1.130",nonce="1489bb04",uri="sip:221@192.168.1.130",response="42bf0a79646c01a0fb4e010b3a51cba1",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
Organization: Doubango Telecom

v=0
o=- 1708632238310997200 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS RjZC3RuRaCy7ZEn7QF97pDs2ZARIy9yBP6Sy
m=audio 56550 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
c=IN IP4 77.87.41.198
a=rtcp:56550 IN IP4 77.87.41.198
a=candidate:4093981320 1 udp 2122260223 192.168.1.130 56550 typ host generation 0
a=candidate:4093981320 2 udp 2122260223 192.168.1.130 56550 typ host generation 0
a=candidate:1967993916 1 udp 1686052607 77.87.41.198 56550 typ srflx raddr 192.168.1.130 rport 56550 generation 0
a=candidate:1967993916 2 udp 1686052607 77.87.41.198 56550 typ srflx raddr 192.168.1.130 rport 56550 generation 0
a=candidate:3129396856 1 tcp 1518280447 192.168.1.130 0 typ host tcptype active generation 0
a=candidate:3129396856 2 tcp 1518280447 192.168.1.130 0 typ host tcptype active generation 0
a=ice-ufrag:S/vjEA1xTE4ilg1Z
a=ice-pwd:P3jwK0NS4JpZ3pe3LF3l+O4j
a=ice-options:google-ice
a=fingerprint:sha-256 29:42:5A:38:C0:3B:8C:D6:F7:60:4F:42:92:E5:7A:51:A3:0C:AB:82:DF:7B:0F:1B:98:E2:2B:B5:D7:C2:0A:84
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:2020025942 cname:F2asousgHTFy3qBT
a=ssrc:2020025942 msid:RjZC3RuRaCy7ZEn7QF97pDs2ZARIy9yBP6Sy 8071dd9b-8fb0-43a5-8969-daac77bf06c2
a=ssrc:2020025942 mslabel:RjZC3RuRaCy7ZEn7QF97pDs2ZARIy9yBP6Sy
a=ssrc:2020025942 label:8071dd9b-8fb0-43a5-8969-daac77bf06c2
<------------->
--- (13 headers 41 lines) ---
Using INVITE request as basis request - ff63e03b-6e45-0131-025f-4acae05c063e
Found peer '222' for '222' from 192.168.1.130:58923
  == Using SIP RTP CoS mark 5
Found RTP audio format 111
Found RTP audio format 103
Found RTP audio format 104
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 106
Found RTP audio format 105
Found RTP audio format 13
Found RTP audio format 126
Found unknown media description format opus for ID 111
Found unknown media description format ISAC for ID 103
Found unknown media description format ISAC for ID 104
Found audio description format G722 for ID 9
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found unknown media description format CN for ID 106
Found unknown media description format CN for ID 105
Found audio description format CN for ID 13
Found audio description format telephone-event for ID 126

<--- Reliably Transmitting (NAT) to 192.168.1.130:58923 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKFBebVekgl49yqdLXslwnmxS6dPccTxft;received=192.168.1.130;rport=58923
From: "222"<sip:222@192.168.1.130:5060>;tag=54hy0vZSvTcIz2AUWGxC
To: <sip:221@192.168.1.130>;tag=as5e7d639f
Call-ID: ff63e03b-6e45-0131-025f-4acae05c063e
CSeq: 8122 INVITE
Server: Asterisk PBX 11.6-cert10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'ff63e03b-6e45-0131-025f-4acae05c063e' in 32000 ms (Method: INVITE)

<--- SIP read from WS:192.168.1.130:58923 --->
ACK sip:221@192.168.1.130 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKFBebVekgl49yqdLXslwnmxS6dPccTxft;rport
From: "222"<sip:222@192.168.1.130:5060>;tag=54hy0vZSvTcIz2AUWGxC
To: <sip:221@192.168.1.130>;tag=as5e7d639f
Call-ID: ff63e03b-6e45-0131-025f-4acae05c063e
CSeq: 8122 ACK
Content-Length: 0
Max-Forwards: 70
Вернуться к началу
ded
Сообщения: 15631
Зарегистрирован: 26 авг 2010, 19:00

Re: Настройка WebRTC Asterisk 11.6

Сообщение ded »

1) Причина:

Код: Выделить всё

SIP response 603 "Failed to get local SDP" back from 192.168.1.130:58842
set_destination: Parsing <sip:222@df7jal23ls0d.invalid;transport=ws> for address/port to send to
set_destination: URI is for WebSocket, we can't set destination
Большую часть этого SDP

Код: Выделить всё

v=0
o=- 1708632238310997200 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS RjZC3RuRaCy7ZEn7QF97pDs2ZARIy9yBP6Sy
m=audio 56550 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
c=IN IP4 77.87.41.198
a=rtcp:56550 IN IP4 77.87.41.198
a=candidate:4093981320 1 udp 2122260223 192.168.1.130 56550 typ host generation 0
a=candidate:4093981320 2 udp 2122260223 192.168.1.130 56550 typ host generation 0
a=candidate:1967993916 1 udp 1686052607 77.87.41.198 56550 typ srflx raddr 192.168.1.130 rport 56550 generation 0
a=candidate:1967993916 2 udp 1686052607 77.87.41.198 56550 typ srflx raddr 192.168.1.130 rport 56550 generation 0
a=candidate:3129396856 1 tcp 1518280447 192.168.1.130 0 typ host tcptype active generation 0
a=candidate:3129396856 2 tcp 1518280447 192.168.1.130 0 typ host tcptype active generation 0
a=ice-ufrag:S/vjEA1xTE4ilg1Z
a=ice-pwd:P3jwK0NS4JpZ3pe3LF3l+O4j
a=ice-options:google-ice
a=fingerprint:sha-256 29:42:5A:38:C0:3B:8C:D6:F7:60:4F:42:92:E5:7A:51:A3:0C:AB:82:DF:7B:0F:1B:98:E2:2B:B5:D7:C2:0A:84
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-h ... -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:2020025942 cname:F2asousgHTFy3qBT
a=ssrc:2020025942 msid:RjZC3RuRaCy7ZEn7QF97pDs2ZARIy9yBP6Sy 8071dd9b-8fb0-43a5-8969-daac77bf06c2
a=ssrc:2020025942 mslabel:RjZC3RuRaCy7ZEn7QF97pDs2ZARIy9yBP6Sy
a=ssrc:2020025942 label:8071dd9b-8fb0-43a5-8969-daac77bf06c2
Астериск не понимает -

Код: Выделить всё

Found RTP audio format 111
Found RTP audio format 103
Found RTP audio format 104
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 106
Found RTP audio format 105
Found RTP audio format 13
Found RTP audio format 126
Found unknown media description format opus for ID 111
Found unknown media description format ISAC for ID 103
Found unknown media description format ISAC for ID 104
Found audio description format G722 for ID 9
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found unknown media description format CN for ID 106
Found unknown media description format CN for ID 105
Found audio description format CN for ID 13
Found audio description format telephone-event for ID 126
Уберите из Dubango неизвестные науке кодеки, и, думаю, пройдёт.

2) Прячьте большие логи под спойлер, please.
Вернуться к началу
aleksandr_star
Сообщения: 17
Зарегистрирован: 24 фев 2015, 13:35

Re: Настройка WebRTC Asterisk 11.6

Сообщение aleksandr_star »

Спасибо. Попробую сделать.
Вернуться к началу
awsswa
Сообщения: 2390
Зарегистрирован: 09 июн 2012, 10:52
Откуда: Россия, Пермь skype: yarick_perm

Re: Настройка WebRTC Asterisk 11.6

Сообщение awsswa »

решение не живучее пока еще на asterisk - при интенсивном использовании рекомендую другие продукты (freeswitch)

ЗЫ кто бы прикрутил вот этот проект http://www.loowid.com - к sip телефонии - пока как конференции видел и лучше, но в свободном доступе пока только это.


Поставил уже в пару мест - народ пользуется
платный суппорт по мере возможностей
Вернуться к началу
Samael28
Сообщения: 1057
Зарегистрирован: 08 янв 2011, 18:32
Откуда: Киев
Контактная информация:
Контактная информация пользователя Samael28

Re: Настройка WebRTC Asterisk 11.6

Сообщение Samael28 »

В варианте связки WebRTC - SIP также рекомендую FreeSwitch(mod_verto). Просто Doubango/sipml/sipjs - это довольно глюкаво
Мой профайл на Upwork
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aleksandr_star
Сообщения: 17
Зарегистрирован: 24 фев 2015, 13:35

Re: Настройка WebRTC Asterisk 11.6

Сообщение aleksandr_star »

Спасибо, всем за советы. Получилось настроить правда на версии Asterisk 13.2
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