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Webrtc куда идёт rtp

Newbies/FAQ Forum 4 сообщений -
#1
Добрый день!

Asterisk 11.2.1 без patch

sip.conf
[8003]
secret=passwo
context=outbound
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
qualify=yes
qualifyfreq=600
transport=udp,wss,ws
encryption=yes
dial=SIP/8003
callerid=Mixa
callcounter=yes
avpf=yes
icesupport=yes
directmedia=no

http.conf

[general]
enabled=yes
bindaddr=0.0.0.0
bindport=8088


Пользуюсь сервисом sipml5, при звонки на номер 8003 (браузер) звонок проходит, голос идёт и туда и обратно, если звонить с браузера, то браузер голос передаёт, но обратно с удалённой стороны голос не слышен
debug:


INVITE sip:7010@tb.tt.ua SIP/2.0
Via: SIP/2.0/UDP 87.106.69.240:11060;branch=z9hG4bKsK0AURRH2tBvUFOqsQiZy7QU7bHRA0p1;rport
From: "8003";tag=owaVhDtZswBcsSkpz919
To:
Contact: "8003";+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: baf1d236-4a03-ecc7-6cd8-2b740d0a9293
CSeq: 60552 INVITE
Content-Type: application/sdp
Content-Length: 2260
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.03.27
Organization: Doubango Telecom
Via: SIP/2.0/TCP 82.117.234.137:35832;rport;branch=z9hG4bKsK0AURRH2tBvUFOqsQiZy7QU7bHRA0p1;ws-hacked=WS

v=0
o=- 2225855495 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS eBoyro4YMopedXLz5qzEjOHBtcxrKzxgI5Ug
m=audio 50319 RTP/SAVPF 103 104 111 0 8 107 106 105 13 126
c=IN IP4 82.117.234.137
a=rtcp:50319 IN IP4 82.117.234.137
a=candidate:2968351779 1 udp 2113937151 192.168.1.211 50319 typ host generation 0
a=candidate:2968351779 2 udp 2113937151 192.168.1.211 50319 typ host generation 0
a=candidate:4062413514 1 udp 2113937151 192.168.122.1 54618 typ host generation 0
a=candidate:4062413514 2 udp 2113937151 192.168.122.1 54618 typ host generation 0
a=candidate:832926359 1 udp 1845501695 82.117.234.137 50319 typ srflx raddr 192.168.1.211 rport 50319 generation 0
a=candidate:832926359 2 udp 1845501695 82.117.234.137 50319 typ srflx raddr 192.168.1.211 rport 50319 generation 0
a=candidate:1936426110 1 udp 1845501695 82.117.234.137 54618 typ srflx raddr 192.168.122.1 rport 54618 generation 0
a=candidate:1936426110 2 udp 1845501695 82.117.234.137 54618 typ srflx raddr 192.168.122.1 rport 54618 generation 0
a=candidate:4268656851 1 tcp 1509957375 192.168.1.211 33766 typ host generation 0
a=candidate:4268656851 2 tcp 1509957375 192.168.1.211 33766 typ host generation 0
a=candidate:3164634682 1 tcp 1509957375 192.168.122.1 57245 typ host generation 0
a=candidate:3164634682 2 tcp 1509957375 192.168.122.1 57245 typ host generation 0
a=ice-ufrag:Go1f2fyvp0PM1wc9
a=ice-pwd:EPC9GKbMkFkaHyGuzSv9N0Er
a=ice-options:google-ice
a=sendrecv
a=mid:audio
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:qT56bgHoFRY9NpW/xYyOnRgjTNvlSsagHUSrwIJn
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:lGbBB3+N58EYeTb+MV6stxsB3dLdDpcqCrYFpWsF
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:107 CN/48000
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:2070840497 cname:AKjerSzsioPH1rw5
a=ssrc:2070840497 msid:eBoyro4YMopedXLz5qzEjOHBtcxrKzxgI5Ug eBoyro4YMopedXLz5qzEjOHBtcxrKzxgI5Uga0
a=ssrc:2070840497 mslabel:eBoyro4YMopedXLz5qzEjOHBtcxrKzxgI5Ug
a=ssrc:2070840497 label:eBoyro4YMopedXLz5qzEjOHBtcxrKzxgI5Uga0

--- (13 headers 45 lines) ---
Sending to 87.106.69.240:11060 (no NAT)
Using INVITE request as basis request - baf1d236-4a03-ecc7-6cd8-2b740d0a9293
Found peer '8003' for '8003' from 87.106.69.240:11060


SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 87.106.69.240:11060;branch=z9hG4bKsK0AURRH2tBvUFOqsQiZy7QU7bHRA0p1;received=87.106.69.240;rport=11060
Via: SIP/2.0/TCP 82.117.234.137:35832;rport;branch=z9hG4bKsK0AURRH2tBvUFOqsQiZy7QU7bHRA0p1;ws-hacked=WS
From: "8003";tag=owaVhDtZswBcsSkpz919
To: ;tag=as7f9d7d37
Call-ID: baf1d236-4a03-ecc7-6cd8-2b740d0a9293
CSeq: 60552 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="33e73995"
Content-Length: 0



Scheduling destruction of SIP dialog 'baf1d236-4a03-ecc7-6cd8-2b740d0a9293' in 6400 ms (Method: INVITE)


ACK sip:7010@tb.tt.ua SIP/2.0
Via: SIP/2.0/UDP 87.106.69.240:11060;branch=z9hG4bKsK0AURRH2tBvUFOqsQiZy7QU7bHRA0p1;rport
From: "8003";tag=owaVhDtZswBcsSkpz919
To: ;tag=as7f9d7d37
Call-ID: baf1d236-4a03-ecc7-6cd8-2b740d0a9293
CSeq: 60552 ACK
Content-Length: 0
Max-Forwards: 70


--- (8 headers 0 lines) ---


ACK sip:7010@tb.tt.ua SIP/2.0
Via: SIP/2.0/UDP 87.106.69.240:11060;branch=z9hG4bKsK0AURRH2tBvUFOqsQiZy7QU7bHRA0p1;rport
From: "8003";tag=owaVhDtZswBcsSkpz919
To: ;tag=as7f9d7d37
Call-ID: baf1d236-4a03-ecc7-6cd8-2b740d0a9293
CSeq: 60552 ACK
Content-Length: 0
Max-Forwards: 70
Via: SIP/2.0/TCP 82.117.234.137:35832;rport;branch=z9hG4bKsK0AURRH2tBvUFOqsQiZy7QU7bHRA0p1;ws-hacked=WS


--- (9 headers 0 lines) ---


INVITE sip:7010@tb.tt.ua SIP/2.0
Via: SIP/2.0/UDP 87.106.69.240:11060;branch=z9hG4bKmHE65q8GHVLXtuLZb1u4BaptqNXMetKo;rport
From: "8003";tag=owaVhDtZswBcsSkpz919
To:
Contact: "8003";+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: baf1d236-4a03-ecc7-6cd8-2b740d0a9293
CSeq: 60553 INVITE
Content-Type: application/sdp
Content-Length: 2260
Max-Forwards: 70
Authorization: Digest username="8003",realm="asterisk",nonce="33e73995",uri="sip:7010@tb.tt.ua",response="5fbb4e70d827efead399234391cd5566",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.03.27
Organization: Doubango Telecom
Via: SIP/2.0/TCP 82.117.234.137:35832;rport;branch=z9hG4bKmHE65q8GHVLXtuLZb1u4BaptqNXMetKo;ws-hacked=WS

v=0
o=- 2225855495 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS eBoyro4YMopedXLz5qzEjOHBtcxrKzxgI5Ug
m=audio 50319 RTP/SAVPF 103 104 111 0 8 107 106 105 13 126
c=IN IP4 82.117.234.137
a=rtcp:50319 IN IP4 82.117.234.137
a=candidate:2968351779 1 udp 2113937151 192.168.1.211 50319 typ host generation 0
a=candidate:2968351779 2 udp 2113937151 192.168.1.211 50319 typ host generation 0
a=candidate:4062413514 1 udp 2113937151 192.168.122.1 54618 typ host generation 0
a=candidate:4062413514 2 udp 2113937151 192.168.122.1 54618 typ host generation 0
a=candidate:832926359 1 udp 1845501695 82.117.234.137 50319 typ srflx raddr 192.168.1.211 rport 50319 generation 0
a=candidate:832926359 2 udp 1845501695 82.117.234.137 50319 typ srflx raddr 192.168.1.211 rport 50319 generation 0
a=candidate:1936426110 1 udp 1845501695 82.117.234.137 54618 typ srflx raddr 192.168.122.1 rport 54618 generation 0
a=candidate:1936426110 2 udp 1845501695 82.117.234.137 54618 typ srflx raddr 192.168.122.1 rport 54618 generation 0
a=candidate:4268656851 1 tcp 1509957375 192.168.1.211 33766 typ host generation 0
a=candidate:4268656851 2 tcp 1509957375 192.168.1.211 33766 typ host generation 0
a=candidate:3164634682 1 tcp 1509957375 192.168.122.1 57245 typ host generation 0
a=candidate:3164634682 2 tcp 1509957375 192.168.122.1 57245 typ host generation 0
a=ice-ufrag:Go1f2fyvp0PM1wc9
a=ice-pwd:EPC9GKbMkFkaHyGuzSv9N0Er
a=ice-options:google-ice
a=sendrecv
a=mid:audio
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:qT56bgHoFRY9NpW/xYyOnRgjTNvlSsagHUSrwIJn
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:lGbBB3+N58EYeTb+MV6stxsB3dLdDpcqCrYFpWsF
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:107 CN/48000
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:2070840497 cname:AKjerSzsioPH1rw5
a=ssrc:2070840497 msid:eBoyro4YMopedXLz5qzEjOHBtcxrKzxgI5Ug eBoyro4YMopedXLz5qzEjOHBtcxrKzxgI5Uga0
a=ssrc:2070840497 mslabel:eBoyro4YMopedXLz5qzEjOHBtcxrKzxgI5Ug
a=ssrc:2070840497 label:eBoyro4YMopedXLz5qzEjOHBtcxrKzxgI5Uga0

--- (14 headers 45 lines) ---
Sending to 87.106.69.240:11060 (no NAT)
Using INVITE request as basis request - baf1d236-4a03-ecc7-6cd8-2b740d0a9293
Found peer '8003' for '8003' from 87.106.69.240:11060
== Using SIP RTP CoS mark 5
Found RTP audio format 103
Found RTP audio format 104
Found RTP audio format 111
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 107
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 ISAC for ID 103
Found unknown media description format ISAC for ID 104
Found unknown media description format opus for ID 111
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found unknown media description format CN for ID 107
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
Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 82.117.234.137:50319
Looking for 7010 in outbound (domain tb.tt.ua)
list_route: hop:


SIP/2.0 100 Trying
Via: SIP/2.0/UDP 87.106.69.240:11060;branch=z9hG4bKmHE65q8GHVLXtuLZb1u4BaptqNXMetKo;received=87.106.69.240;rport=11060
Via: SIP/2.0/TCP 82.117.234.137:35832;rport;branch=z9hG4bKmHE65q8GHVLXtuLZb1u4BaptqNXMetKo;ws-hacked=WS
From: "8003";tag=owaVhDtZswBcsSkpz919
To:
Call-ID: baf1d236-4a03-ecc7-6cd8-2b740d0a9293
CSeq: 60553 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Length: 0



-- Executing [7010@outbound:1] Wait("SIP/8003-000034a2", "1") in new stack


ACK sip:7010@tb.tt.ua SIP/2.0
Via: SIP/2.0/UDP 87.106.69.240:11060;branch=z9hG4bKsK0AURRH2tBvUFOqsQiZy7QU7bHRA0p1;rport
From: "8003";tag=owaVhDtZswBcsSkpz919
To: ;tag=as7f9d7d37
Call-ID: baf1d236-4a03-ecc7-6cd8-2b740d0a9293
CSeq: 60552 ACK
Content-Length: 0
Max-Forwards: 70
Via: SIP/2.0/TCP 82.117.234.137:35832;rport;branch=z9hG4bKsK0AURRH2tBvUFOqsQiZy7QU7bHRA0p1;ws-hacked=WS


--- (9 headers 0 lines) ---
-- Executing [7010@outbound:2] Wait("SIP/8003-000034a2", "1") in new stack


ACK sip:7010@tb.tt.ua SIP/2.0
Via: SIP/2.0/UDP 87.106.69.240:11060;branch=z9hG4bKsK0AURRH2tBvUFOqsQiZy7QU7bHRA0p1;rport
From: "8003";tag=owaVhDtZswBcsSkpz919
To: ;tag=as7f9d7d37
Call-ID: baf1d236-4a03-ecc7-6cd8-2b740d0a9293
CSeq: 60552 ACK
Content-Length: 0
Max-Forwards: 70
Via: SIP/2.0/TCP 82.117.234.137:35832;rport;branch=z9hG4bKsK0AURRH2tBvUFOqsQiZy7QU7bHRA0p1;ws-hacked=WS


--- (9 headers 0 lines) ---
-- Executing [7010@outbound:3] Set("SIP/8003-000034a2", "fname=1365780555.13475") in new stack
-- Executing [7010@outbound:4] MixMonitor("SIP/8003-000034a2", "/var/spool/asterisk/monitor/20130412/1365780555.13475.wav") in new stack
-- Executing [7010@outbound:5] Dial("SIP/8003-000034a2", "SIP/7010") in new stack
== Begin MixMonitor Recording SIP/8003-000034a2
== Using SIP RTP CoS mark 5
-- Called SIP/7010
-- SIP/7010-000034a3 is ringing


SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 87.106.69.240:11060;branch=z9hG4bKmHE65q8GHVLXtuLZb1u4BaptqNXMetKo;received=87.106.69.240;rport=11060
Via: SIP/2.0/TCP 82.117.234.137:35832;rport;branch=z9hG4bKmHE65q8GHVLXtuLZb1u4BaptqNXMetKo;ws-hacked=WS
From: "8003";tag=owaVhDtZswBcsSkpz919
To: ;tag=as63c4b618
Call-ID: baf1d236-4a03-ecc7-6cd8-2b740d0a9293
CSeq: 60553 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Length: 0





ACK sip:7010@tb.tt.ua SIP/2.0
Via: SIP/2.0/UDP 87.106.69.240:11060;branch=z9hG4bKsK0AURRH2tBvUFOqsQiZy7QU7bHRA0p1;rport
From: "8003";tag=owaVhDtZswBcsSkpz919
To: ;tag=as7f9d7d37
Call-ID: baf1d236-4a03-ecc7-6cd8-2b740d0a9293
CSeq: 60552 ACK
Content-Length: 0
Max-Forwards: 70
Via: SIP/2.0/TCP 82.117.234.137:35832;rport;branch=z9hG4bKsK0AURRH2tBvUFOqsQiZy7QU7bHRA0p1;ws-hacked=WS


--- (9 headers 0 lines) ---
-- SIP/7010-000034a3 answered SIP/8003-000034a2
Audio is at 18254
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP


SIP/2.0 200 OK
Via: SIP/2.0/UDP 87.106.69.240:11060;branch=z9hG4bKmHE65q8GHVLXtuLZb1u4BaptqNXMetKo;received=87.106.69.240;rport=11060
Via: SIP/2.0/TCP 82.117.234.137:35832;rport;branch=z9hG4bKmHE65q8GHVLXtuLZb1u4BaptqNXMetKo;ws-hacked=WS
From: "8003";tag=owaVhDtZswBcsSkpz919
To: ;tag=as63c4b618
Call-ID: baf1d236-4a03-ecc7-6cd8-2b740d0a9293
CSeq: 60553 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 598

v=0
o=root 1550017834 1550017834 IN IP4 89.184.65.55
s=Asterisk PBX 11.2.1
c=IN IP4 89.184.65.55
t=0 0
m=audio 18254 RTP/SAVPF 0 8 101
a=rtpmap:0 PCMU/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:53fd3271616c65345d525e492a2e8f46
a=ice-pwd:0efd29384932990417bb9e85551fb970
a=candidate:H59b84137 1 UDP 2130706431 89.184.65.55 18254 typ host
a=candidate:H59b84137 2 UDP 2130706430 89.184.65.55 18255 typ host
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:UkpRfrZ3yY777gsPmtGb8FKQlXUHe+Z6YIXahl6o


Retransmitting #1 (no NAT) to 87.106.69.240:11060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 87.106.69.240:11060;branch=z9hG4bKmHE65q8GHVLXtuLZb1u4BaptqNXMetKo;received=87.106.69.240;rport=11060
Via: SIP/2.0/TCP 82.117.234.137:35832;rport;branch=z9hG4bKmHE65q8GHVLXtuLZb1u4BaptqNXMetKo;ws-hacked=WS
From: "8003";tag=owaVhDtZswBcsSkpz919
To: ;tag=as63c4b618
Call-ID: baf1d236-4a03-ecc7-6cd8-2b740d0a9293
CSeq: 60553 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 598

v=0
o=root 1550017834 1550017834 IN IP4 89.184.65.55
s=Asterisk PBX 11.2.1
c=IN IP4 89.184.65.55
t=0 0
m=audio 18254 RTP/SAVPF 0 8 101
a=rtpmap:0 PCMU/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:53fd3271616c65345d525e492a2e8f46
a=ice-pwd:0efd29384932990417bb9e85551fb970
a=candidate:H59b84137 1 UDP 2130706431 89.184.65.55 18254 typ host
a=candidate:H59b84137 2 UDP 2130706430 89.184.65.55 18255 typ host
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:UkpRfrZ3yY777gsPmtGb8FKQlXUHe+Z6YIXahl6o

---


ACK sip:7010@89.184.65.55:5060 SIP/2.0
Via: SIP/2.0/UDP 87.106.69.240:11060;branch=z9hG4bKFTZDgDIMfbDzxsJQhqXT;rport
From: "8003";tag=owaVhDtZswBcsSkpz919
To: ;tag=as63c4b618
Contact: "8003";+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: baf1d236-4a03-ecc7-6cd8-2b740d0a9293
CSeq: 60553 ACK
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="8003",realm="asterisk",nonce="33e73995",uri="sip:7010@89.184.65.55:5060",response="da0f81fdf5dbb20a3eefadb7dac957dd",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.03.27
Organization: Doubango Telecom
Via: SIP/2.0/TCP 82.117.234.137:35832;rport;branch=z9hG4bKFTZDgDIMfbDzxsJQhqXT;ws-hacked=WS


--- (13 headers 0 lines) ---


ACK sip:7010@89.184.65.55:5060 SIP/2.0
Via: SIP/2.0/UDP 87.106.69.240:11060;branch=z9hG4bKCMFkbsnslfVFxciHEPoy;rport
From: "8003";tag=owaVhDtZswBcsSkpz919
To: ;tag=as63c4b618
Contact: "8003";+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: baf1d236-4a03-ecc7-6cd8-2b740d0a9293
CSeq: 60553 ACK
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="8003",realm="asterisk",nonce="33e73995",uri="sip:7010@89.184.65.55:5060",response="da0f81fdf5dbb20a3eefadb7dac957dd",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.03.27
Organization: Doubango Telecom
Via: SIP/2.0/TCP 82.117.234.137:35832;rport;branch=z9hG4bKCMFkbsnslfVFxciHEPoy;ws-hacked=WS


--- (13 headers 0 lines) ---


ACK sip:7010@89.184.65.55:5060 SIP/2.0
Via: SIP/2.0/UDP 87.106.69.240:11060;branch=z9hG4bKFTZDgDIMfbDzxsJQhqXT;rport
From: "8003";tag=owaVhDtZswBcsSkpz919
To: ;tag=as63c4b618
Contact: "8003";+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: baf1d236-4a03-ecc7-6cd8-2b740d0a9293
CSeq: 60553 ACK
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="8003",realm="asterisk",nonce="33e73995",uri="sip:7010@89.184.65.55:5060",response="da0f81fdf5dbb20a3eefadb7dac957dd",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.03.27
Organization: Doubango Telecom
Via: SIP/2.0/TCP 82.117.234.137:35832;rport;branch=z9hG4bKFTZDgDIMfbDzxsJQhqXT;ws-hacked=WS


--- (13 headers 0 lines) ---


ACK sip:7010@89.184.65.55:5060 SIP/2.0
Via: SIP/2.0/UDP 87.106.69.240:11060;branch=z9hG4bKCMFkbsnslfVFxciHEPoy;rport
From: "8003";tag=owaVhDtZswBcsSkpz919
To: ;tag=as63c4b618
Contact: "8003";+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: baf1d236-4a03-ecc7-6cd8-2b740d0a9293
CSeq: 60553 ACK
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="8003",realm="asterisk",nonce="33e73995",uri="sip:7010@89.184.65.55:5060",response="da0f81fdf5dbb20a3eefadb7dac957dd",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.03.27
Organization: Doubango Telecom
Via: SIP/2.0/TCP 82.117.234.137:35832;rport;branch=z9hG4bKCMFkbsnslfVFxciHEPoy;ws-hacked=WS


--- (13 headers 0 lines) ---


ACK sip:7010@89.184.65.55:5060 SIP/2.0
Via: SIP/2.0/UDP 87.106.69.240:11060;branch=z9hG4bKFTZDgDIMfbDzxsJQhqXT;rport
From: "8003";tag=owaVhDtZswBcsSkpz919
To: ;tag=as63c4b618
Contact: "8003";+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: baf1d236-4a03-ecc7-6cd8-2b740d0a9293
CSeq: 60553 ACK
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="8003",realm="asterisk",nonce="33e73995",uri="sip:7010@89.184.65.55:5060",response="da0f81fdf5dbb20a3eefadb7dac957dd",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.03.27
Organization: Doubango Telecom
Via: SIP/2.0/TCP 82.117.234.137:35832;rport;branch=z9hG4bKFTZDgDIMfbDzxsJQhqXT;ws-hacked=WS


--- (13 headers 0 lines) ---


ACK sip:7010@89.184.65.55:5060 SIP/2.0
Via: SIP/2.0/UDP 87.106.69.240:11060;branch=z9hG4bKCMFkbsnslfVFxciHEPoy;rport
From: "8003";tag=owaVhDtZswBcsSkpz919
To: ;tag=as63c4b618
Contact: "8003";+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: baf1d236-4a03-ecc7-6cd8-2b740d0a9293
CSeq: 60553 ACK
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="8003",realm="asterisk",nonce="33e73995",uri="sip:7010@89.184.65.55:5060",response="da0f81fdf5dbb20a3eefadb7dac957dd",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.03.27
Organization: Doubango Telecom
Via: SIP/2.0/TCP 82.117.234.137:35832;rport;branch=z9hG4bKCMFkbsnslfVFxciHEPoy;ws-hacked=WS


--- (13 headers 0 lines) ---
== Spawn extension (outbound, 7010, 5) exited non-zero on 'SIP/8003-000034a2'
Scheduling destruction of SIP dialog 'baf1d236-4a03-ecc7-6cd8-2b740d0a9293' in 6400 ms (Method: ACK)
set_destination: Parsing for address/port to send to
set_destination: set destination to 87.106.69.240:11060
Reliably Transmitting (no NAT) to 87.106.69.240:11060:
BYE sip:8003@87.106.69.240:11060;rtcweb-breaker=no;click2call=no;transport=udp;ws-src-ip=82.117.234.137;ws-src-port=35832;ws-src-proto=ws SIP/2.0
Via: SIP/2.0/UDP 89.184.65.55:5060;branch=z9hG4bK19ef2a20;rport
Max-Forwards: 70
From: ;tag=as63c4b618
To: "8003";tag=owaVhDtZswBcsSkpz919
Call-ID: baf1d236-4a03-ecc7-6cd8-2b740d0a9293
CSeq: 102 BYE
User-Agent: Asterisk PBX 11.2.1
Proxy-Authorization: Digest username="8003", realm="asterisk", algorithm=MD5, uri="sip:tb.tt.ua", nonce="33e73995", response="ac081e502d2b2b707f0349bda222d663"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/8003-000034a2


SIP/2.0 200 OK
Via: SIP/2.0/UDP 89.184.65.55:5060;rport=5060;branch=z9hG4bK19ef2a20
From: ;tag=as63c4b618
To: "8003";tag=owaVhDtZswBcsSkpz919
Contact:
Call-ID: baf1d236-4a03-ecc7-6cd8-2b740d0a9293
CSeq: 102 BYE
Content-Length: 0


--- (8 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog 'baf1d236-4a03-ecc7-6cd8-2b740d0a9293' Method: ACK


REGISTER sip:tb.tt.ua SIP/2.0
Via: SIP/2.0/UDP 87.106.69.240:11060;branch=z9hG4bK5LVoggsFBPUiTpNgdtZnfry0BfBOmXby;rport
From: "8003";tag=C6VQIpL2w2IOszcOCHow
To: "8003"
Contact: "8003";expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: d539fb08-3cce-847b-c150-fca077b22b19
CSeq: 23275 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="8003",realm="asterisk",nonce="7d063cd5",uri="sip:tb.tt.ua",response="adae1583ef7d1cd16b5c56db17295325",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.03.27
Organization: Doubango Telecom
Via: SIP/2.0/TCP 82.117.234.137:35832;rport;branch=z9hG4bK5LVoggsFBPUiTpNgdtZnfry0BfBOmXby;ws-hacked=WS


--- (13 headers 0 lines) ---
Sending to 87.106.69.240:11060 (no NAT)


SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 87.106.69.240:11060;branch=z9hG4bK5LVoggsFBPUiTpNgdtZnfry0BfBOmXby;received=87.106.69.240;rport=11060
Via: SIP/2.0/TCP 82.117.234.137:35832;rport;branch=z9hG4bK5LVoggsFBPUiTpNgdtZnfry0BfBOmXby;ws-hacked=WS
From: "8003";tag=C6VQIpL2w2IOszcOCHow
To: "8003";tag=as033a5e12
Call-ID: d539fb08-3cce-847b-c150-fca077b22b19
CSeq: 23275 REGISTER
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3bfe97ca"
Content-Length: 0



Scheduling destruction of SIP dialog 'd539fb08-3cce-847b-c150-fca077b22b19' in 32000 ms (Method: REGISTER)


REGISTER sip:tb.tt.ua SIP/2.0
Via: SIP/2.0/UDP 87.106.69.240:11060;branch=z9hG4bK4NEEq5SCaRMZy1W1FNqXfn1XELd0S2tQ;rport
From: "8003";tag=C6VQIpL2w2IOszcOCHow
To: "8003"
Contact: "8003";expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: d539fb08-3cce-847b-c150-fca077b22b19
CSeq: 23276 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="8003",realm="asterisk",nonce="3bfe97ca",uri="sip:tb.tt.ua",response="d873bd1d07c36c73329e5588f190aaf7",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.03.27
Organization: Doubango Telecom
Via: SIP/2.0/TCP 82.117.234.137:35832;rport;branch=z9hG4bK4NEEq5SCaRMZy1W1FNqXfn1XELd0S2tQ;ws-hacked=WS


--- (13 headers 0 lines) ---
Sending to 87.106.69.240:11060 (no NAT)
Reliably Transmitting (no NAT) to 87.106.69.240:11060:
OPTIONS sip:8003@87.106.69.240:11060;rtcweb-breaker=no;transport=udp;ws-src-ip=82.117.234.137;ws-src-port=35832;ws-src-proto=ws SIP/2.0
Via: SIP/2.0/UDP 89.184.65.55:5060;branch=z9hG4bK001045b3
Max-Forwards: 70
From: "asterisk" ;tag=as5e250b3c
To:
Contact:
Call-ID: 4d009c28713e7782287e916e0f35dfd7@89.184.65.55:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.2.1
Date: Fri, 12 Apr 2013 15:29:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---


SIP/2.0 200 OK
Via: SIP/2.0/UDP 87.106.69.240:11060;branch=z9hG4bK4NEEq5SCaRMZy1W1FNqXfn1XELd0S2tQ;received=87.106.69.240;rport=11060
Via: SIP/2.0/TCP 82.117.234.137:35832;rport;branch=z9hG4bK4NEEq5SCaRMZy1W1FNqXfn1XELd0S2tQ;ws-hacked=WS
From: "8003";tag=C6VQIpL2w2IOszcOCHow
To: "8003";tag=as033a5e12
Call-ID: d539fb08-3cce-847b-c150-fca077b22b19
CSeq: 23276 REGISTER
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 200
Contact: ;expires=200
Date: Fri, 12 Apr 2013 15:29:23 GMT
Content-Length: 0



Scheduling destruction of SIP dialog 'd539fb08-3cce-847b-c150-fca077b22b19' in 32000 ms (Method: REGISTER)


SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/UDP 89.184.65.55:5060;branch=z9hG4bK001045b3
From: "asterisk";tag=as5e250b3c
To:
Call-ID: 4d009c28713e7782287e916e0f35dfd7@89.184.65.55:5060
CSeq: 102 OPTIONS
Content-Length: 0


--- (7 headers 0 lines) ---
Really destroying SIP dialog '4d009c28713e7782287e916e0f35dfd7@89.184.65.55:5060' Method: OPTIONS


ACK sip:7010@tb.tt.ua SIP/2.0
Via: SIP/2.0/UDP 87.106.69.240:11060;branch=z9hG4bKsK0AURRH2tBvUFOqsQiZy7QU7bHRA0p1;rport
From: "8003";tag=owaVhDtZswBcsSkpz919
To: ;tag=as7f9d7d37
Call-ID: baf1d236-4a03-ecc7-6cd8-2b740d0a9293
CSeq: 60552 ACK
Content-Length: 0
Max-Forwards: 70
Via: SIP/2.0/TCP 82.117.234.137:35832;rport;branch=z9hG4bKsK0AURRH2tBvUFOqsQiZy7QU7bHRA0p1;ws-hacked=WS


--- (9 headers 0 lines) ---


Возможно кто-то уже встречался с данной проблемой?
#2
У меня Asterisk 11.4.0 без патчей

Конфиги аналогичные. Если звонить в браузер, то звонящего слышно, а звонящий не слышит ничего, если звонить с браузера, то голос ходит нормально в обе стороны. Иногда когда звонишь в браузер голос все таки проходит, т.е раз на раз не приходится, даже не знаю хуже это или нет.

Сравнивал логи обоих случаев, для астериска все идентично.

Кто нить решил этот вопрос ?
#3
11.4 тоже без патчей.

Звонок из браузера - все ок. А вот в браузер даже вызов не идет, не то что голос Sad
_________________
P4 3.0 + 1Gb CentOS 5.8 Aster 1.8.16
Не люблю gui-сборки: натуральный продукт вкуснее.
И еще: я ПРОФИ так как НЕ ЛЕНЮСЬ читать литературу.
#4
Что-то пишет в логах ?

У меня проблема в том, что Иногда оно работает, вот это хуже, если б просто не работало было бы наверное проще.

PS Сейчас сижу на 11.5.1 без патчей