Advertisement
Not a member of Pastebin yet?
Sign Up,
it unlocks many cool features!
- Asterisk 11.11.0, 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.
- =========================================================================
- Connected to Asterisk 11.11.0 currently running on asterisk (pid = 2599)
- asterisk*CLI>
- <--- SIP read from WS:10.0.1.107:51114 --->
- INVITE sip:011525555555555@asterisk SIP/2.0
- Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKvq67CGtbT7RG8w2ZOTeyLTChBsM1aM5m;rport
- From: "WEBRTC"<sip:5005@10.0.1.108>;tag=4D1TKAr7hDF8WhKcmaoZ
- To: <sip:011525555555555@asterisk>
- Contact: "WEBRTC"<sip:5005@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>
- Call-ID: 36bb3d83-bb35-d00f-829f-4180a329a7fa
- CSeq: 35986 INVITE
- Content-Type: application/sdp
- Content-Length: 1835
- Route: <sip:10.0.1.108:5060;lr;sipml5-outbound;transport=udp>
- ax-Forwards: 70
- User-Agent: DM_SIPWEB-UA
- Organization: Digital-Merge
- v=0
- o=- 6498442466790256000 2 IN IP4 127.0.0.1
- s=Doubango Telecom - chrome
- t=0 0
- a=group:BUNDLE audio
- a=msid-semantic: WMS t7qGTnlmAnKdVBPKvjVvRUEXAL4mUeASMy0B
- m=audio 60478 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126
- c=IN IP4 192.168.56.1
- a=rtcp:60478 IN IP4 192.168.56.1
- a=candidate:2999745851 1 udp 2122260223 192.168.56.1 60478 typ host generation 0
- a=candidate:2999745851 2 udp 2122260223 192.168.56.1 60478 typ host generation 0
- a=candidate:2322994768 1 udp 2122194687 10.0.1.107 60479 typ host generation 0
- a=candidate:2322994768 2 udp 2122194687 10.0.1.107 60479 typ host generation 0
- a=candidate:4233069003 1 tcp 1518280447 192.168.56.1 0 typ host generation 0
- a=candidate:4233069003 2 tcp 1518280447 192.168.56.1 0 typ host generation 0
- a=candidate:3304450720 1 tcp 1518214911 10.0.1.107 0 typ host generation 0
- a=candidate:3304450720 2 tcp 1518214911 10.0.1.107 0 typ host generation 0
- a=ice-ufrag:Z2alDX+HHwpejZTd
- a=ice-pwd:V6i3HRU3ygTpQ4ETnNHEJ8xS
- a=ice-options:google-ice
- a=fingerprint:sha-256 D5:C5:EB:62:8F:E8:C4:D5:6A:C6:EF:2F:65:0C:6B:6B:01:31:FA:42:CD:24:CF:8C:30:98:90:61:E4:59:6B:DD
- 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: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:573627479 cname:clMeURHWmYKPuarq
- a=ssrc:573627479 msid:t7qGTnlmAnKdVBPKvjVvRUEXAL4mUeASMy0B 3950e683-b0aa-41ed-869f-b3df7dc1066c
- a=ssrc:573627479 mslabel:t7qGTnlmAnKdVBPKvjVvRUEXAL4mUeASMy0B
- a=ssrc:573627479 label:3950e683-b0aa-41ed-869f-b3df7dc1066c
- <------------->
- asterisk*CLI>
- --- (13 headers 42 lines) ---
- asterisk*CLI>
- Using INVITE request as basis request - 36bb3d83-bb35-d00f-829f-4180a329a7fa
- asterisk*CLI>
- Found peer '5005' for '5005' from 10.0.1.107:51114
- asterisk*CLI>
- <--- Reliably Transmitting (NAT) to 10.0.1.107:51114 --->
- SIP/2.0 401 Unauthorized
- Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKvq67CGtbT7RG8w2ZOTeyLTChBsM1aM5m;received=10.0.1.107;rport=51114
- From: "WEBRTC"<sip:5005@10.0.1.108>;tag=4D1TKAr7hDF8WhKcmaoZ
- To: <sip:011525555555555@asterisk>;tag=as52ef68de
- Call-ID: 36bb3d83-bb35-d00f-829f-4180a329a7fa
- CSeq: 35986 INVITE
- Server: Digital-Merge_UA
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="774b7649"
- Content-Length: 0
- <------------>
- asterisk*CLI>
- Scheduling destruction of SIP dialog '36bb3d83-bb35-d00f-829f-4180a329a7fa' in 8000 ms (Method: INVITE)
- asterisk*CLI>
- <--- SIP read from WS:10.0.1.107:51114 --->
- ACK sip:011525555555555@asterisk SIP/2.0
- Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKvq67CGtbT7RG8w2ZOTeyLTChBsM1aM5m;rport
- From: "WEBRTC"<sip:5005@10.0.1.108>;tag=4D1TKAr7hDF8WhKcmaoZ
- To: <sip:011525555555555@asterisk>;tag=as52ef68de
- Call-ID: 36bb3d83-bb35-d00f-829f-4180a329a7fa
- CSeq: 35986 ACK
- Content-Length: 0
- Route: <sip:10.0.1.108:5060;lr;sipml5-outbound;transport=udp>
- ax-Forwards: 70
- <------------->
- asterisk*CLI>
- --- (9 headers 0 lines) ---
- asterisk*CLI>
- <--- SIP read from WS:10.0.1.107:51114 --->
- INVITE sip:011525555555555@asterisk SIP/2.0
- Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKGSGhfHFVUWyo1OsHmyQgdcxL3NFhI9lz;rport
- From: "WEBRTC"<sip:5005@10.0.1.108>;tag=4D1TKAr7hDF8WhKcmaoZ
- To: <sip:011525555555555@asterisk>
- Contact: "WEBRTC"<sip:5005@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>
- Call-ID: 36bb3d83-bb35-d00f-829f-4180a329a7fa
- CSeq: 35987 INVITE
- Content-Type: application/sdp
- Content-Length: 1835
- Route: <sip:10.0.1.108:5060;lr;sipml5-outbound;transport=udp>
- ax-Forwards: 70
- Authorization: Digest username="5005",realm="asterisk",nonce="774b7649",uri="sip:011525555555555@asterisk",response="0cf7ef57b65ad0a96f0c2dc0855074ac",algorithm=MD5
- User-Agent: DM_SIPWEB-UA
- Organization: Digital-Merge
- v=0
- o=- 6498442466790256000 2 IN IP4 127.0.0.1
- s=Doubango Telecom - chrome
- t=0 0
- a=group:BUNDLE audio
- a=msid-semantic: WMS t7qGTnlmAnKdVBPKvjVvRUEXAL4mUeASMy0B
- m=audio 60478 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126
- c=IN IP4 192.168.56.1
- a=rtcp:60478 IN IP4 192.168.56.1
- a=candidate:2999745851 1 udp 2122260223 192.168.56.1 60478 typ host generation 0
- a=candidate:2999745851 2 udp 2122260223 192.168.56.1 60478 typ host generation 0
- a=candidate:2322994768 1 udp 2122194687 10.0.1.107 60479 typ host generation 0
- a=candidate:2322994768 2 udp 2122194687 10.0.1.107 60479 typ host generation 0
- a=candidate:4233069003 1 tcp 1518280447 192.168.56.1 0 typ host generation 0
- a=candidate:4233069003 2 tcp 1518280447 192.168.56.1 0 typ host generation 0
- a=candidate:3304450720 1 tcp 1518214911 10.0.1.107 0 typ host generation 0
- a=candidate:3304450720 2 tcp 1518214911 10.0.1.107 0 typ host generation 0
- a=ice-ufrag:Z2alDX+HHwpejZTd
- a=ice-pwd:V6i3HRU3ygTpQ4ETnNHEJ8xS
- a=ice-options:google-ice
- a=fingerprint:sha-256 D5:C5:EB:62:8F:E8:C4:D5:6A:C6:EF:2F:65:0C:6B:6B:01:31:FA:42:CD:24:CF:8C:30:98:90:61:E4:59:6B:DD
- 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: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:573627479 cname:clMeURHWmYKPuarq
- a=ssrc:573627479 msid:t7qGTnlmAnKdVBPKvjVvRUEXAL4mUeASMy0B 3950e683-b0aa-41ed-869f-b3df7dc1066c
- a=ssrc:573627479 mslabel:t7qGTnlmAnKdVBPKvjVvRUEXAL4mUeASMy0B
- a=ssrc:573627479 label:3950e683-b0aa-41ed-869f-b3df7dc1066c
- <------------->
- --- (14 headers 42 lines) ---
- Using INVITE request as basis request - 36bb3d83-bb35-d00f-829f-4180a329a7fa
- Found peer '5005' for '5005' from 10.0.1.107:51114
- asterisk*CLI>
- == Using SIP RTP CoS mark 5
- asterisk*CLI>
- Found RTP audio format 111
- asterisk*CLI>
- Found RTP audio format 103
- asterisk*CLI>
- Found RTP audio format 104
- asterisk*CLI>
- Found RTP audio format 0
- asterisk*CLI>
- Found RTP audio format 8
- asterisk*CLI>
- Found RTP audio format 106
- asterisk*CLI>
- Found RTP audio format 105
- asterisk*CLI>
- Found RTP audio format 13
- asterisk*CLI>
- Found RTP audio format 126
- asterisk*CLI>
- Found unknown media description format opus for ID 111
- asterisk*CLI>
- Found unknown media description format ISAC for ID 103
- asterisk*CLI>
- Found unknown media description format ISAC for ID 104
- asterisk*CLI>
- 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
- asterisk*CLI>
- 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
- asterisk*CLI>
- Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
- asterisk*CLI>
- Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)
- asterisk*CLI>
- Peer audio RTP is at port 192.168.56.1:60478
- asterisk*CLI>
- Looking for 011525555555555 in wrtc (domain asterisk)
- asterisk*CLI>
- list_route: hop: <sip:5005@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>
- asterisk*CLI>
- <--- Transmitting (NAT) to 10.0.1.107:51114 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKGSGhfHFVUWyo1OsHmyQgdcxL3NFhI9lz;received=10.0.1.107;rport=51114
- From: "WEBRTC"<sip:5005@10.0.1.108>;tag=4D1TKAr7hDF8WhKcmaoZ
- To: <sip:011525555555555@asterisk>
- Call-ID: 36bb3d83-bb35-d00f-829f-4180a329a7fa
- CSeq: 35987 INVITE
- Server: Digital-Merge_UA
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- Contact: <sip:011525555555555@10.0.1.108:5060;transport=WS>
- Content-Length: 0
- <------------>
- -- Executing [011525555555555@wrtc:1] Dial("SIP/5005-0000000d", "SIP/MySuperSIPProvider/011525555555555,40,TtWw") in new stack
- asterisk*CLI>
- == Using SIP RTP CoS mark 5
- asterisk*CLI>
- Audio is at 20974
- asterisk*CLI>
- Adding codec 100003 (ulaw) to SDP
- asterisk*CLI>
- Adding codec 100002 (gsm) to SDP
- asterisk*CLI>
- Adding codec 100004 (alaw) to SDP
- asterisk*CLI>
- Adding non-codec 0x1 (telephone-event) to SDP
- asterisk*CLI>
- Reliably Transmitting (NAT) to 74.54.XXX.XXX:5060:
- INVITE sip:011525555555555@dallas.voip.ms SIP/2.0
- Via: SIP/2.0/UDP 10.0.1.108:5060;branch=z9hG4bK53edb567;rport
- ax-Forwards: 70
- From: "WebRTC" <sip:143987@10.0.1.108>;tag=as613a93d5
- To: <sip:011525555555555@dallas.voip.ms>
- Contact: <sip:143987@10.0.1.108:5060>
- Call-ID: 703b63e61c82683c5487fe2f66792dc3@10.0.1.108:5060
- CSeq: 102 INVITE
- User-Agent: Digital-Merge_UA
- Date: Sat, 02 Aug 2014 17:58:19 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- Remote-Party-ID: "WebRTC" <sip:5005@10.0.1.108>;party=calling;privacy=off;screen=no
- Content-Type: application/sdp
- Content-Length: 279
- v=0
- o=root 1175191988 1175191988 IN IP4 10.0.1.108
- s=Asterisk PBX 11.11.0
- c=IN IP4 10.0.1.108
- t=0 0
- m=audio 20974 RTP/AVP 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=ptime:20
- a=sendrecv
- ---
- -- Called SIP/MySuperSIPProvider/011525555555555
- asterisk*CLI>
- Retransmitting #1 (NAT) to 74.54.XXX.XXX:5060:
- INVITE sip:011525555555555@dallas.voip.ms SIP/2.0
- Via: SIP/2.0/UDP 10.0.1.108:5060;branch=z9hG4bK53edb567;rport
- ax-Forwards: 70
- From: "WebRTC" <sip:143987@10.0.1.108>;tag=as613a93d5
- To: <sip:011525555555555@dallas.voip.ms>
- Contact: <sip:143987@10.0.1.108:5060>
- Call-ID: 703b63e61c82683c5487fe2f66792dc3@10.0.1.108:5060
- CSeq: 102 INVITE
- User-Agent: Digital-Merge_UA
- Date: Sat, 02 Aug 2014 17:58:19 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- Remote-Party-ID: "WebRTC" <sip:5005@10.0.1.108>;party=calling;privacy=off;screen=no
- Content-Type: application/sdp
- Content-Length: 279
- v=0
- o=root 1175191988 1175191988 IN IP4 10.0.1.108
- s=Asterisk PBX 11.11.0
- c=IN IP4 10.0.1.108
- t=0 0
- m=audio 20974 RTP/AVP 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=ptime:20
- a=sendrecv
- ---
- asterisk*CLI>
- Retransmitting #2 (NAT) to 74.54.XXX.XXX:5060:
- INVITE sip:011525555555555@dallas.voip.ms SIP/2.0
- Via: SIP/2.0/UDP 10.0.1.108:5060;branch=z9hG4bK53edb567;rport
- ax-Forwards: 70
- From: "WebRTC" <sip:143987@10.0.1.108>;tag=as613a93d5
- To: <sip:011525555555555@dallas.voip.ms>
- Contact: <sip:143987@10.0.1.108:5060>
- Call-ID: 703b63e61c82683c5487fe2f66792dc3@10.0.1.108:5060
- CSeq: 102 INVITE
- User-Agent: Digital-Merge_UA
- Date: Sat, 02 Aug 2014 17:58:19 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- Remote-Party-ID: "WebRTC" <sip:5005@10.0.1.108>;party=calling;privacy=off;screen=no
- Content-Type: application/sdp
- Content-Length: 279
- v=0
- o=root 1175191988 1175191988 IN IP4 10.0.1.108
- s=Asterisk PBX 11.11.0
- c=IN IP4 10.0.1.108
- t=0 0
- m=audio 20974 RTP/AVP 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=ptime:20
- a=sendrecv
- ---
- asterisk*CLI>
- <--- SIP read from UDP:74.54.XXX.XXX:5060 --->
- SIP/2.0 407 Proxy Authentication Required
- Via: SIP/2.0/UDP 10.0.1.108:5060;branch=z9hG4bK53edb567;received=189.241.6.199;rport=13027
- From: "WebRTC" <sip:143987@10.0.1.108>;tag=as613a93d5
- To: <sip:011525555555555@dallas.voip.ms>;tag=as52d6980b
- Call-ID: 703b63e61c82683c5487fe2f66792dc3@10.0.1.108:5060
- CSeq: 102 INVITE
- User-Agent: voip.ms
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Proxy-Authenticate: Digest algorithm=MD5, realm="dallas.voip.ms", nonce="795f186d"
- Content-Length: 0
- <------------->
- asterisk*CLI>
- --- (11 headers 0 lines) ---
- asterisk*CLI>
- Transmitting (NAT) to 74.54.XXX.XXX:5060:
- ACK sip:011525555555555@dallas.voip.ms SIP/2.0
- Via: SIP/2.0/UDP 10.0.1.108:5060;branch=z9hG4bK53edb567;rport
- ax-Forwards: 70
- From: "WebRTC" <sip:143987@10.0.1.108>;tag=as613a93d5
- To: <sip:011525555555555@dallas.voip.ms>;tag=as52d6980b
- Contact: <sip:143987@10.0.1.108:5060>
- Call-ID: 703b63e61c82683c5487fe2f66792dc3@10.0.1.108:5060
- CSeq: 102 ACK
- User-Agent: Digital-Merge_UA
- Content-Length: 0
- ---
- asterisk*CLI>
- Audio is at 20974
- asterisk*CLI>
- Adding codec 100003 (ulaw) to SDP
- Adding codec 100002 (gsm) to SDP
- Adding codec 100004 (alaw) to SDP
- Adding non-codec 0x1 (telephone-event) to SDP
- asterisk*CLI>
- Reliably Transmitting (NAT) to 74.54.XXX.XXX:5060:
- INVITE sip:011525555555555@dallas.voip.ms SIP/2.0
- Via: SIP/2.0/UDP 10.0.1.108:5060;branch=z9hG4bK404698f6;rport
- ax-Forwards: 70
- From: "WebRTC" <sip:143987@10.0.1.108>;tag=as613a93d5
- To: <sip:011525555555555@dallas.voip.ms>
- Contact: <sip:143987@10.0.1.108:5060>
- Call-ID: 703b63e61c82683c5487fe2f66792dc3@10.0.1.108:5060
- CSeq: 103 INVITE
- User-Agent: Digital-Merge_UA
- Proxy-Authorization: Digest username="143987", realm="dallas.voip.ms", algorithm=MD5, uri="sip:011525555555555@dallas.voip.ms", nonce="795f186d", response="d8bf9497de2ffae33a1e09729300a777"
- Date: Sat, 02 Aug 2014 17:58:20 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- Remote-Party-ID: "WebRTC" <sip:5005@10.0.1.108>;party=calling;privacy=off;screen=no
- Content-Type: application/sdp
- Content-Length: 279
- v=0
- o=root 1175191988 1175191989 IN IP4 10.0.1.108
- s=Asterisk PBX 11.11.0
- c=IN IP4 10.0.1.108
- t=0 0
- m=audio 20974 RTP/AVP 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=ptime:20
- a=sendrecv
- ---
- <--- SIP read from UDP:74.54.XXX.XXX:5060 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/UDP 10.0.1.108:5060;branch=z9hG4bK404698f6;received=189.241.6.199;rport=13027
- From: "WebRTC" <sip:143987@10.0.1.108>;tag=as613a93d5
- To: <sip:011525555555555@dallas.voip.ms>
- Call-ID: 703b63e61c82683c5487fe2f66792dc3@10.0.1.108:5060
- CSeq: 103 INVITE
- User-Agent: voip.ms
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Contact: <sip:011525555555555@74.54.XXX.XXX>
- Content-Length: 0
- <------------->
- --- (11 headers 0 lines) ---
- asterisk*CLI>
- <--- SIP read from UDP:74.54.XXX.XXX:5060 --->
- SIP/2.0 183 Session Progress
- Via: SIP/2.0/UDP 10.0.1.108:5060;branch=z9hG4bK404698f6;received=189.241.6.199;rport=13027
- From: "WebRTC" <sip:143987@10.0.1.108>;tag=as613a93d5
- To: <sip:011525555555555@dallas.voip.ms>;tag=as15352569
- Call-ID: 703b63e61c82683c5487fe2f66792dc3@10.0.1.108:5060
- CSeq: 103 INVITE
- User-Agent: voip.ms
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Contact: <sip:011525555555555@74.54.XXX.XXX>
- Content-Type: application/sdp
- Content-Length: 263
- v=0
- o=root 18731 18731 IN IP4 74.54.XXX.XXX
- s=session
- c=IN IP4 74.54.XXX.XXX
- t=0 0
- m=audio 18042 RTP/AVP 0 3 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:3 GSM/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=silenceSupp:off - - - -
- a=ptime:20
- a=sendrecv
- <------------->
- asterisk*CLI>
- --- (12 headers 13 lines) ---
- asterisk*CLI>
- list_route: hop: <sip:011525555555555@74.54.XXX.XXX>
- Found RTP audio format 0
- Found RTP audio format 3
- Found RTP audio format 101
- Found audio description format PCMU for ID 0
- Found audio description format GSM for ID 3
- Found audio description format telephone-event for ID 101
- Capabilities: us - (gsm|ulaw|alaw), peer - audio=(gsm|ulaw)/video=(nothing)/text=(nothing), combined - (gsm|ulaw)
- Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
- Peer audio RTP is at port 74.54.XXX.XXX:18042
- -- SIP/MySuperSIPProvider-0000000e is making progress passing it to SIP/5005-0000000d
- asterisk*CLI>
- Audio is at 19880
- asterisk*CLI>
- Adding codec 100003 (ulaw) to SDP
- asterisk*CLI>
- Adding codec 100004 (alaw) to SDP
- asterisk*CLI>
- Adding non-codec 0x1 (telephone-event) to SDP
- asterisk*CLI>
- <--- Transmitting (NAT) to 10.0.1.107:51114 --->
- SIP/2.0 183 Session Progress
- Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKGSGhfHFVUWyo1OsHmyQgdcxL3NFhI9lz;received=10.0.1.107;rport=51114
- From: "WEBRTC"<sip:5005@10.0.1.108>;tag=4D1TKAr7hDF8WhKcmaoZ
- To: <sip:011525555555555@asterisk>;tag=as654c7257
- Call-ID: 36bb3d83-bb35-d00f-829f-4180a329a7fa
- CSeq: 35987 INVITE
- Server: Digital-Merge_UA
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- Contact: <sip:011525555555555@10.0.1.108:5060;transport=WS>
- Content-Type: application/sdp
- Content-Length: 777
- v=0
- o=root 462328627 462328627 IN IP4 10.0.1.108
- s=Asterisk PBX 11.11.0
- c=IN IP4 10.0.1.108
- t=0 0
- m=audio 19880 UDP/TLS/RTP/SAVPF 0 8 126
- a=rtpmap:0 PCMU/8000
- a=rtpmap:8 PCMA/8000
- a=rtpmap:126 telephone-event/8000
- a=fmtp:126 0-16
- a=ptime:20
- a=ice-ufrag:4693399c4251f1db40f8054a159138d7
- a=ice-pwd:0d4f0a8861da5d277fd5f466337a088d
- a=candidate:Ha00016c 1 UDP 2130706431 10.0.1.108 19880 typ host
- a=candidate:Sbdf106c7 1 UDP 1694498815 189.241.6.199 13082 typ srflx
- a=candidate:Ha00016c 2 UDP 2130706430 10.0.1.108 19881 typ host
- a=candidate:Sbdf106c7 2 UDP 1694498814 189.241.6.199 13084 typ srflx
- a=connection:new
- a=setup:active
- a=fingerprint:SHA-256 49:BF:9C:44:4B:E8:63:28:31:3C:36:7D:7C:F9:DC:6A:C8:AF:71:C0:E3:3D:36:E0:87:C0:27:00:9E:FC:FC:6A
- a=sendrecv
- <------------>
- asterisk*CLI>
- > 0xb7511138 -- Probation passed - setting RTP source address to 10.0.1.107:60479
- asterisk*CLI>
- > 0xb7511138 -- Probation passed - setting RTP source address to 10.0.1.107:60479
- asterisk*CLI>
- > 0xb7382f18 -- Probation passed - setting RTP source address to 74.54.XXX.XXX:18042
- asterisk*CLI>
- <--- SIP read from UDP:74.54.XXX.XXX:5060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 10.0.1.108:5060;branch=z9hG4bK404698f6;received=189.241.6.199;rport=13027
- From: "WebRTC" <sip:143987@10.0.1.108>;tag=as613a93d5
- To: <sip:011525555555555@dallas.voip.ms>;tag=as15352569
- Call-ID: 703b63e61c82683c5487fe2f66792dc3@10.0.1.108:5060
- CSeq: 103 INVITE
- User-Agent: voip.ms
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Contact: <sip:011525555555555@74.54.XXX.XXX>
- Content-Type: application/sdp
- Content-Length: 263
- v=0
- o=root 18731 18732 IN IP4 74.54.XXX.XXX
- s=session
- c=IN IP4 74.54.XXX.XXX
- t=0 0
- m=audio 18042 RTP/AVP 0 3 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:3 GSM/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=silenceSupp:off - - - -
- a=ptime:20
- a=sendrecv
- <------------->
- asterisk*CLI>
- --- (12 headers 13 lines) ---
- asterisk*CLI>
- Found RTP audio format 0
- asterisk*CLI>
- Found RTP audio format 3
- Found RTP audio format 101
- asterisk*CLI>
- Found audio description format PCMU for ID 0
- Found audio description format GSM for ID 3
- Found audio description format telephone-event for ID 101
- Capabilities: us - (gsm|ulaw|alaw), peer - audio=(gsm|ulaw)/video=(nothing)/text=(nothing), combined - (gsm|ulaw)
- Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
- asterisk*CLI>
- Peer audio RTP is at port 74.54.XXX.XXX:18042
- list_route: hop: <sip:011525555555555@74.54.XXX.XXX>
- set_destination: Parsing <sip:011525555555555@74.54.XXX.XXX> for address/port to send to
- set_destination: set destination to 74.54.XXX.XXX:5060
- Transmitting (NAT) to 74.54.XXX.XXX:5060:
- ACK sip:011525555555555@74.54.XXX.XXX SIP/2.0
- Via: SIP/2.0/UDP 10.0.1.108:5060;branch=z9hG4bK0152723a;rport
- ax-Forwards: 70
- From: "WebRTC" <sip:143987@10.0.1.108>;tag=as613a93d5
- To: <sip:011525555555555@dallas.voip.ms>;tag=as15352569
- Contact: <sip:143987@10.0.1.108:5060>
- Call-ID: 703b63e61c82683c5487fe2f66792dc3@10.0.1.108:5060
- CSeq: 103 ACK
- User-Agent: Digital-Merge_UA
- Content-Length: 0
- ---
- -- SIP/MySuperSIPProvider-0000000e answered SIP/5005-0000000d
- Audio is at 19880
- Adding codec 100003 (ulaw) to SDP
- Adding codec 100004 (alaw) to SDP
- Adding non-codec 0x1 (telephone-event) to SDP
- <--- Reliably Transmitting (NAT) to 10.0.1.107:51114 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKGSGhfHFVUWyo1OsHmyQgdcxL3NFhI9lz;received=10.0.1.107;rport=51114
- From: "WEBRTC"<sip:5005@10.0.1.108>;tag=4D1TKAr7hDF8WhKcmaoZ
- To: <sip:011525555555555@asterisk>;tag=as654c7257
- Call-ID: 36bb3d83-bb35-d00f-829f-4180a329a7fa
- CSeq: 35987 INVITE
- Server: Digital-Merge_UA
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- Contact: <sip:011525555555555@10.0.1.108:5060;transport=WS>
- Content-Type: application/sdp
- Content-Length: 782
- v=0
- o=root 462328627 462328627 IN IP4 10.0.1.108
- s=Asterisk PBX 11.11.0
- c=IN IP4 10.0.1.108
- t=0 0
- m=audio 19880 UDP/TLS/RTP/SAVPF 0 8 126
- a=rtpmap:0 PCMU/8000
- a=rtpmap:8 PCMA/8000
- a=rtpmap:126 telephone-event/8000
- a=fmtp:126 0-16
- a=ptime:20
- a=ice-ufrag:4693399c4251f1db40f8054a159138d7
- a=ice-pwd:0d4f0a8861da5d277fd5f466337a088d
- a=candidate:Ha00016c 1 UDP 2130706431 10.0.1.108 19880 typ host
- a=candidate:Sbdf106c7 1 UDP 1694498815 189.241.6.199 13082 typ srflx
- a=candidate:Ha00016c 2 UDP 2130706430 10.0.1.108 19881 typ host
- a=candidate:Sbdf106c7 2 UDP 1694498814 189.241.6.199 13084 typ srflx
- a=connection:existing
- a=setup:active
- a=fingerprint:SHA-256 49:BF:9C:44:4B:E8:63:28:31:3C:36:7D:7C:F9:DC:6A:C8:AF:71:C0:E3:3D:36:E0:87:C0:27:00:9E:FC:FC:6A
- a=sendrecv
- <------------>
- asterisk*CLI>
- > 0xb7382f18 -- Probation passed - setting RTP source address to 74.54.XXX.XXX:18042
- asterisk*CLI>
- <--- SIP read from WS:10.0.1.107:51114 --->
- ACK sip:011525555555555@10.0.1.108:5060;transport=WS SIP/2.0
- Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKMrQxdOwCdr9pIGWDZPg4;rport
- From: "WEBRTC"<sip:5005@10.0.1.108>;tag=4D1TKAr7hDF8WhKcmaoZ
- To: <sip:011525555555555@asterisk>;tag=as654c7257
- Contact: "WEBRTC"<sip:5005@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>
- Call-ID: 36bb3d83-bb35-d00f-829f-4180a329a7fa
- CSeq: 35987 ACK
- Content-Length: 0
- Route: <sip:10.0.1.108:5060;lr;sipml5-outbound;transport=udp>
- ax-Forwards: 70
- Authorization: Digest username="5005",realm="asterisk",nonce="774b7649",uri="sip:011525555555555@10.0.1.108:5060;transport=WS",response="ddb5a7829e26a98edbe09483987c894d",algorithm=MD5
- User-Agent: DM_SIPWEB-UA
- Organization: Digital-Merge
- <------------->
- --- (13 headers 0 lines) ---
- asterisk*CLI> rtp set debug on
- asterisk*CLI> RTP Debugging Enabled
- asterisk*CLI>
- Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039707, ts 3166686328, len 000033)
- Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033589, ts 3166686328, len 000170)
- asterisk*CLI>
- Got RTP packet from 10.0.1.107:60479 (type 00, seq 018367, ts 1324339245, len 000160)
- asterisk*CLI>
- Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041587, ts 1324339240, len 000033)
- asterisk*CLI>
- Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039708, ts 3166686488, len 000033)
- asterisk*CLI>
- Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033590, ts 3166686488, len 000170)
- asterisk*CLI>
- Got RTP packet from 10.0.1.107:60479 (type 00, seq 018368, ts 1324339405, len 000160)
- Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041588, ts 1324339400, len 000033)
- asterisk*CLI>
- Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039709, ts 3166686648, len 000033)
- Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033591, ts 3166686648, len 000170)
- asterisk*CLI>
- Got RTP packet from 10.0.1.107:60479 (type 00, seq 018369, ts 1324339565, len 000160)
- asterisk*CLI>
- Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041589, ts 1324339560, len 000033)
- asterisk*CLI>
- Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039710, ts 3166686808, len 000033)
- asterisk*CLI>
- Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033592, ts 3166686808, len 000170)
- asterisk*CLI>
- Got RTP packet from 10.0.1.107:60479 (type 00, seq 018370, ts 1324339725, len 000160)
- asterisk*CLI>
- Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041590, ts 1324339720, len 000033)
- asterisk*CLI>
- Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039711, ts 3166686968, len 000033)
- asterisk*CLI>
- Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033593, ts 3166686968, len 000170)
- asterisk*CLI>
- Got RTP packet from 10.0.1.107:60479 (type 00, seq 018371, ts 1324339885, len 000160)
- Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041591, ts 1324339880, len 000033)
- Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039712, ts 3166687128, len 000033)
- asterisk*CLI>
- Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033594, ts 3166687128, len 000170)
- asterisk*CLI>
- Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039713, ts 3166687288, len 000033)
- asterisk*CLI>
- Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033595, ts 3166687288, len 000170)
- asterisk*CLI>
- Got RTP packet from 10.0.1.107:60479 (type 00, seq 018372, ts 1324340045, len 000160)
- asterisk*CLI>
- Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041592, ts 1324340040, len 000033)
- asterisk*CLI>
- Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039714, ts 3166687448, len 000033)
- asterisk*CLI>
- Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033596, ts 3166687448, len 000170)
- asterisk*CLI>
- Got RTP packet from 10.0.1.107:60479 (type 00, seq 018373, ts 1324340205, len 000160)
- asterisk*CLI>
- Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041593, ts 1324340200, len 000033)
- asterisk*CLI>
- Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039715, ts 3166687608, len 000033)
- asterisk*CLI>
- Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033597, ts 3166687608, len 000170)
- asterisk*CLI>
- Got RTP packet from 10.0.1.107:60479 (type 00, seq 018374, ts 1324340365, len 000160)
- asterisk*CLI>
- Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041594, ts 1324340360, len 000033)
- asterisk*CLI>
- Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039716, ts 3166687768, len 000033)
- asterisk*CLI>
- Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033598, ts 3166687768, len 000170)
- asterisk*CLI>
- Got RTP packet from 10.0.1.107:60479 (type 00, seq 018375, ts 1324340525, len 000160)
- asterisk*CLI>
- Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041595, ts 1324340520, len 000033)
- asterisk*CLI>
- Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039717, ts 3166687928, len 000033)
- Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033599, ts 3166687928, len 000170)
- asterisk*CLI>
- Got RTP packet from 10.0.1.107:60479 (type 00, seq 018376, ts 1324340685, len 000160)
- Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041596, ts 1324340680, len 000033)
- Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039718, ts 3166688088, len 000033)
- asterisk*CLI>
- Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033600, ts 3166688088, len 000170)
- Got RTP packet from 10.0.1.107:60479 (type 00, seq 018377, ts 1324340845, len 000160)
- asterisk*CLI>
- Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041597, ts 1324340840, len 000033)
- asterisk*CLI>
- Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039719, ts 3166688248, len 000033)
- asterisk*CLI>
- Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033601, ts 3166688248, len 000170)
- asterisk*CLI>
- Got RTP packet from 10.0.1.107:60479 (type 00, seq 018378, ts 1324341005, len 000160)
- asterisk*CLI> rtp set debug on
- Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041598, ts 1324341000, len 000033)
- asterisk*CLI> rtp set debug on
- Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039720, ts 3166688408, len 000033)
- asterisk*CLI> rtp set debug on
- Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033602, ts 3166688408, len 000170)
- asterisk*CLI> rtp set debug on
- Got RTP packet from 10.0.1.107:60479 (type 00, seq 018379, ts 1324341165, len 000160)
- asterisk*CLI> rtp set debug on
- Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041599, ts 1324341160, len 000033)
- asterisk*CLI> rtp set debug on
- Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039721, ts 3166688568, len 000033)
- asterisk*CLI> rtp set debug on
- Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033603, ts 3166688568, len 000170)
- asterisk*CLI> rtp set debug on
- Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039722, ts 3166688728, len 000033)
- asterisk*CLI> rtp set debug on
- Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033604, ts 3166688728, len 000170)
- asterisk*CLI> rtp set debug on
- Got RTP packet from 10.0.1.107:60479 (type 00, seq 018380, ts 1324341325, len 000160)
- asterisk*CLI> rtp set debug on
- Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041600, ts 1324341320, len 000033)
- asterisk*CLI> rtp set debug on
- Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039723, ts 3166688888, len 000033)
- asterisk*CLI> rtp set debug on
- Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033605, ts 3166688888, len 000170)
- asterisk*CLI> rtp set debug on
- Got RTP packet from 10.0.1.107:60479 (type 00, seq 018381, ts 1324341485, len 000160)
- asterisk*CLI> rtp set debug on
- Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041601, ts 1324341480, len 000033)
- asterisk*CLI> rtp set debug on
- Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039724, ts 3166689048, len 000033)
- asterisk*CLI> rtp set debug on
- Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033606, ts 3166689048, len 000170)
- asterisk*CLI> rtp set debug on
- Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039725, ts 3166689208, len 000033)
- asterisk*CLI> rtp set debug on
- Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033607, ts 3166689208, len 000170)
- asterisk*CLI> rtp set debug on
- Got RTP packet from 10.0.1.107:60479 (type 00, seq 018382, ts 1324341645, len 000160)
- asterisk*CLI> rtp set debug on
- Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041602, ts 1324341640, len 000033)
- asterisk*CLI> rtp set debug on
- Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039726, ts 3166689368, len 000033)
- asterisk*CLI> rtp set debug on
- Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033608, ts 3166689368, len 000170)
- asterisk*CLI> rtp set debug on
- Got RTP packet from 10.0.1.107:60479 (type 00, seq 018383, ts 1324341805, len 000160)
- asterisk*CLI> rtp set debug on
- Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041603, ts 1324341800, len 000033)
- asterisk*CLI> rtp set debug on
- Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039727, ts 3166689528, len 000033)
- asterisk*CLI> rtp set debug on
- Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033609, ts 3166689528, len 000170)
- Got RTP packet from 10.0.1.107:60479 (type 00, seq 018384, ts 1324341965, len 000160)
- Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041604, ts 1324341960, len 000033)
- Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039728, ts 3166689688, len 000033)
- Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033610, ts 3166689688, len 000170)
- Got RTP packet from 10.0.1.107:60479 (type 00, seq 018385, ts 1324342125, len 000160)
- asterisk*CLI> rtp set debug on
- Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041605, ts 1324342120, len 000033)
- asterisk*CLI> rtp set debug on
- Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039729, ts 3166689848, len 000033)
- asterisk*CLI> rtp set debug on
- Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033611, ts 3166689848, len 000170)
- Got RTP packet from 10.0.1.107:60479 (type 00, seq 018386, ts 1324342285, len 000160)
- Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041606, ts 1324342280, len 000033)
- Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039730, ts 3166690008, len 000033)
- asterisk*CLI> rtp set debug on
- Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033612, ts 3166690008, len 000170)
- asterisk*CLI> rtp set debug on
- Got RTP packet from 10.0.1.107:60479 (type 00, seq 018387, ts 1324342445, len 000160)
- asterisk*CLI> rtp set debug on
- Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041607, ts 1324342440, len 000033)
- asterisk*CLI> rtp set debug on
- Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039731, ts 3166690168, len 000033)
- asterisk*CLI> rtp set debug on
- Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033613, ts 3166690168, len 000170)
- asterisk*CLI> rtp set debug on
- Got RTP packet from 10.0.1.107:60479 (type 00, seq 018388, ts 1324342605, len 000160)
- asterisk*CLI> rtp set debug on
- Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041608, ts 1324342600, len 000033)
- asterisk*CLI> rtp set debug on
- Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039732, ts 3166690328, len 000033)
- asterisk*CLI> rtp set debug on
- Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033614, ts 3166690328, len 000170)
- asterisk*CLI> rtp set debug on
- Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039733, ts 3166690488, len 000033)
- asterisk*CLI> rtp set debug on
- Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033615, ts 3166690488, len 000170)
- asterisk*CLI> rtp set debug o
- Got RTP packet from 10.0.1.107:60479 (type 00, seq 018389, ts 1324342765, len 000160)
- asterisk*CLI> rtp set debug o
- Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041609, ts 1324342760, len 000033)
- asterisk*CLI> rtp set debug o
- Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039734, ts 3166690648, len 000033)
- asterisk*CLI> rtp set debug o
- Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033616, ts 3166690648, len 000170)
- asterisk*CLI> rtp set debug o
- Got RTP packet from 10.0.1.107:60479 (type 00, seq 018390, ts 1324342925, len 000160)
- Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041610, ts 1324342920, len 000033)
- asterisk*CLI> rtp set debug o
- Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039735, ts 3166690808, len 000033)
- asterisk*CLI> rtp set debug o
- Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033617, ts 3166690808, len 000170)
- Got RTP packet from 10.0.1.107:60479 (type 00, seq 018391, ts 1324343085, len 000160)
- Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041611, ts 1324343080, len 000033)
- Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039736, ts 3166690968, len 000033)
- asterisk*CLI> rtp set debug o
- Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033618, ts 3166690968, len 000170)
- asterisk*CLI> rtp set debug o
- Got RTP packet from 10.0.1.107:60479 (type 00, seq 018392, ts 1324343245, len 000160)
- asterisk*CLI> rtp set debug o
- Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041612, ts 1324343240, len 000033)
- Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039737, ts 3166691128, len 000033)
- Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033619, ts 3166691128, len 000170)
- Got RTP packet from 10.0.1.107:60479 (type 00, seq 018393, ts 1324343405, len 000160)
- asterisk*CLI> rtp set debug o
- Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041613, ts 1324343400, len 000033)
- Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039738, ts 3166691288, len 000033)
- Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033620, ts 3166691288, len 000170)
- Got RTP packet from 10.0.1.107:60479 (type 00, seq 018394, ts 1324343565, len 000160)
- Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041614, ts 1324343560, len 000033)
- Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039739, ts 3166691448, len 000033)
- Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033621, ts 3166691448, len 000170)
- Got RTP packet from 10.0.1.107:60479 (type 00, seq 018395, ts 1324343725, len 000160)
- Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041615, ts 1324343720, len 000033)
- Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039740, ts 3166691608, len 000033)
- Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033622, ts 3166691608, len 000170)
- Got RTP packet from 10.0.1.107:60479 (type 00, seq 018396, ts 1324343885, len 000160)
- Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041616, ts 1324343880, len 000033)
- Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039741, ts 3166691768, len 000033)
- Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033623, ts 3166691768, len 000170)
- Got RTP packet from 10.0.1.107:60479 (type 00, seq 018397, ts 1324344045, len 000160)
- Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041617, ts 1324344040, len 000033)
- Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039742, ts 3166691928, len 000033)
- asterisk*CLI> rtp set debug o
- Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033624, ts 3166691928, len 000170)
- asterisk*CLI> rtp set debug o
- Got RTP packet from 10.0.1.107:60479 (type 00, seq 018398, ts 1324344205, len 000160)
- asterisk*CLI> rtp set debug o
- Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041618, ts 1324344200, len 000033)
- Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039743, ts 3166692088, len 000033)
- asterisk*CLI> rtp set debug o
- Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033625, ts 3166692088, len 000170)
- asterisk*CLI> rtp set debug o
- Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039744, ts 3166692248, len 000033)
- asterisk*CLI> rtp set debug o
- Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033626, ts 3166692248, len 000170)
- asterisk*CLI> rtp set debug o
- Got RTP packet from 10.0.1.107:60479 (type 00, seq 018399, ts 1324344365, len 000160)
- asterisk*CLI> rtp set debug o
- Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041619, ts 1324344360, len 000033)
- asterisk*CLI> rtp set debug of
- Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039745, ts 3166692408, len 000033)
- asterisk*CLI> rtp set debug of
- Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033627, ts 3166692408, len 000170)
- asterisk*CLI> rtp set debug of
- Got RTP packet from 10.0.1.107:60479 (type 00, seq 018400, ts 1324344525, len 000160)
- asterisk*CLI> rtp set debug of
- Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041620, ts 1324344520, len 000033)
- Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039746, ts 3166692568, len 000033)
- asterisk*CLI> rtp set debug off
- Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033628, ts 3166692568, len 000170)
- asterisk*CLI> rtp set debug off
- Got RTP packet from 10.0.1.107:60479 (type 00, seq 018401, ts 1324344685, len 000160)
- asterisk*CLI> rtp set debug off
- Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041621, ts 1324344680, len 000033)
- asterisk*CLI> rtp set debug off
- Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039747, ts 3166692728, len 000033)
- asterisk*CLI> rtp set debug off
- Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033629, ts 3166692728, len 000170)
- asterisk*CLI> rtp set debug off
- Got RTP packet from 10.0.1.107:60479 (type 00, seq 018402, ts 1324344845, len 000160)
- asterisk*CLI> rtp set debug off
- asterisk*CLI> Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041622, ts 1324344840, len 000033)
- asterisk*CLI>
- RTP Debugging Disabled
- asterisk*CLI>
- Got RTP packet from 74.54.XXX.XXX:18042 (type 03, seq 039748, ts 3166692888, len 000033)
- Sent RTP packet to 10.0.1.107:60479 (via ICE) (type 00, seq 033630, ts 3166692888, len 000170)
- Got RTP packet from 10.0.1.107:60479 (type 00, seq 018403, ts 1324345005, len 000160)
- Sent RTP packet to 74.54.XXX.XXX:18042 (type 03, seq 041623, ts 1324345000, len 000033)
- asterisk*CLI>
- <--- SIP read from WS:10.0.1.107:51114 --->
- BYE sip:011525555555555@10.0.1.108:5060;transport=WS SIP/2.0
- Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKOfenJK1FwIZ55oAdVTQgUV0XUiugfgmJ;rport
- From: "WEBRTC"<sip:5005@10.0.1.108>;tag=4D1TKAr7hDF8WhKcmaoZ
- To: <sip:011525555555555@asterisk>;tag=as654c7257
- Call-ID: 36bb3d83-bb35-d00f-829f-4180a329a7fa
- CSeq: 35988 BYE
- Content-Length: 0
- Route: <sip:10.0.1.108:5060;lr;sipml5-outbound;transport=udp>
- ax-Forwards: 70
- Authorization: Digest username="5005",realm="asterisk",nonce="774b7649",uri="sip:011525555555555@10.0.1.108:5060;transport=WS",response="db4ddf767da6689802886efbdcbdc01b",algorithm=MD5
- User-Agent: DM_SIPWEB-UA
- Organization: Digital-Merge
- <------------->
- --- (12 headers 0 lines) ---
- Scheduling destruction of SIP dialog '36bb3d83-bb35-d00f-829f-4180a329a7fa' in 8000 ms (Method: BYE)
- asterisk*CLI>
- <--- Transmitting (NAT) to 10.0.1.107:51114 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKOfenJK1FwIZ55oAdVTQgUV0XUiugfgmJ;received=10.0.1.107;rport=51114
- From: "WEBRTC"<sip:5005@10.0.1.108>;tag=4D1TKAr7hDF8WhKcmaoZ
- To: <sip:011525555555555@asterisk>;tag=as654c7257
- Call-ID: 36bb3d83-bb35-d00f-829f-4180a329a7fa
- CSeq: 35988 BYE
- Server: Digital-Merge_UA
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- Content-Length: 0
- <------------>
- asterisk*CLI>
- Scheduling destruction of SIP dialog '703b63e61c82683c5487fe2f66792dc3@10.0.1.108:5060' in 6400 ms (Method: INVITE)
- asterisk*CLI>
- set_destination: Parsing <sip:011525555555555@74.54.XXX.XXX> for address/port to send to
- asterisk*CLI>
- set_destination: set destination to 74.54.XXX.XXX:5060
- Reliably Transmitting (NAT) to 74.54.XXX.XXX:5060:
- BYE sip:011525555555555@74.54.XXX.XXX SIP/2.0
- Via: SIP/2.0/UDP 10.0.1.108:5060;branch=z9hG4bK49190ac8;rport
- ax-Forwards: 70
- From: "WebRTC" <sip:143987@10.0.1.108>;tag=as613a93d5
- To: <sip:011525555555555@dallas.voip.ms>;tag=as15352569
- Call-ID: 703b63e61c82683c5487fe2f66792dc3@10.0.1.108:5060
- CSeq: 104 BYE
- User-Agent: Digital-Merge_UA
- Proxy-Authorization: Digest username="143987", realm="dallas.voip.ms", algorithm=MD5, uri="sip:011525555555555@74.54.XXX.XXX", nonce="795f186d", response="74894d08f01c5106099e34bfbf63666a"
- X-Asterisk-HangupCause: Normal Clearing
- X-Asterisk-HangupCauseCode: 16
- Content-Length: 0
- ---
- == Spawn extension (wrtc, 011525555555555, 1) exited non-zero on 'SIP/5005-0000000d'
- asterisk*CLI>
- <--- SIP read from UDP:74.54.XXX.XXX:5060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 10.0.1.108:5060;branch=z9hG4bK49190ac8;received=189.241.6.199;rport=13027
- From: "WebRTC" <sip:143987@189.241.6.199:13027>;tag=as613a93d5
- To: <sip:011525555555555@dallas.voip.ms>;tag=as15352569
- Call-ID: 703b63e61c82683c5487fe2f66792dc3@10.0.1.108:5060
- CSeq: 104 BYE
- User-Agent: voip.ms
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Content-Length: 0
- <------------->
- --- (10 headers 0 lines) ---
- asterisk*CLI>
- Really destroying SIP dialog '703b63e61c82683c5487fe2f66792dc3@10.0.1.108:5060' Method: INVITE
- asterisk*CLI>
- Reliably Transmitting (NAT) to 74.54.XXX.XXX:5060:
- OPTIONS sip:dallas.voip.ms SIP/2.0
- Via: SIP/2.0/UDP 10.0.1.108:5060;branch=z9hG4bK77f75df0;rport
- ax-Forwards: 70
- From: "asterisk" <sip:143987@10.0.1.108>;tag=as05dccb4b
- To: <sip:dallas.voip.ms>
- Contact: <sip:143987@10.0.1.108:5060>
- Call-ID: 5e08b96b0b65d76775b8aa0810f92ed5@10.0.1.108:5060
- CSeq: 102 OPTIONS
- User-Agent: Digital-Merge_UA
- Date: Sat, 02 Aug 2014 17:59:04 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- Content-Length: 0
- ---
- asterisk*CLI>
- <--- SIP read from UDP:74.54.XXX.XXX:5060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 10.0.1.108:5060;branch=z9hG4bK77f75df0;received=189.241.6.199;rport=13027
- From: "asterisk" <sip:143987@10.0.1.108>;tag=as05dccb4b
- To: <sip:dallas.voip.ms>;tag=as167f2ced
- Call-ID: 5e08b96b0b65d76775b8aa0810f92ed5@10.0.1.108:5060
- CSeq: 102 OPTIONS
- User-Agent: voip.ms
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Contact: <sip:74.54.XXX.XXX>
- Accept: application/sdp
- Content-Length: 0
- <------------->
- asterisk*CLI>
- --- (12 headers 0 lines) ---
- asterisk*CLI>
- Really destroying SIP dialog '5e08b96b0b65d76775b8aa0810f92ed5@10.0.1.108:5060' Method: OPTIONS
- asterisk*CLI>
- Reliably Transmitting (NAT) to 10.0.1.107:51114:
- OPTIONS sip:5005@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws SIP/2.0
- Via: SIP/2.0/WS 10.0.1.108:5060;branch=z9hG4bK09ea7b59;rport
- ax-Forwards: 70
- From: "asterisk" <sip:asterisk@10.0.1.108>;tag=as7d80fe23
- To: <sip:5005@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>
- Contact: <sip:asterisk@10.0.1.108:5060;transport=WS>
- Call-ID: 2710f14a340cc2f535d62e751b17d405@10.0.1.108:5060
- CSeq: 102 OPTIONS
- User-Agent: Digital-Merge_UA
- Date: Sat, 02 Aug 2014 17:59:04 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- Content-Length: 0
- ---
- asterisk*CLI>
- <--- SIP read from WS:10.0.1.107:51114 --->
- SIP/2.0 405 Method Not Allowed
- Via: SIP/2.0/WS 10.0.1.108:5060;rport=5060;branch=z9hG4bK09ea7b59
- From: "asterisk"<sip:asterisk@10.0.1.108>;tag=as7d80fe23
- To: <sip:5005@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>
- Call-ID: 2710f14a340cc2f535d62e751b17d405@10.0.1.108:5060
- CSeq: 102 OPTIONS
- Content-Length: 0
- <------------->
- asterisk*CLI>
- --- (7 headers 0 lines) ---
- asterisk*CLI>
- Really destroying SIP dialog '2710f14a340cc2f535d62e751b17d405@10.0.1.108:5060' Method: OPTIONS
- asterisk*CLI>
Advertisement
Add Comment
Please, Sign In to add comment
Advertisement