[asterisk-bugs] [JIRA] (ASTERISK-25024) there is no ice when i make originale
Joshua Colp (JIRA)
noreply at issues.asterisk.org
Tue Apr 28 15:21:33 CDT 2015
[ https://issues.asterisk.org/jira/browse/ASTERISK-25024?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=226041#comment-226041 ]
Joshua Colp commented on ASTERISK-25024:
----------------------------------------
We appreciate the difficulties you are facing, however this does not appear to be a bug report and your request or comments would be better served in a different forum.
The Asterisk community provides support over IRC, mailing lists, and forums as described at http://asterisk.org/community. The Asterisk issue tracker is used specifically to track issues concerning bugs and documentation errors.
Please see the Asterisk Issue Guidelines [1] for instruction on the intended use of the Asterisk issue tracker.
Thanks!
[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines
> there is no ice when i make originale
> -------------------------------------
>
> Key: ASTERISK-25024
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-25024
> Project: Asterisk
> Issue Type: Information Request
> Security Level: None
> Affects Versions: 11.15.0
> Reporter: Anton Satskiy
> Severity: Minor
>
> 1-when i am doing call from webrtc i get ice working
> {noformat}
> <--- SIP read from WS:91.196.158.205:1466 --->
> INVITE sip:0669197533 at 77.91.132.9 SIP/2.0
> Via: SIP/2.0/WS 7cvtd9ihs2e8.invalid;branch=z9hG4bK8749315
> Max-Forwards: 69
> To: <sip:0669197533 at 77.91.132.9>
> From: "Anton" <sip:1065 at 77.91.132.9>;tag=5i21qaop43
> Call-ID: ocq4hu8eol3kijsgvt6b
> CSeq: 1465 INVITE
> Authorization: Digest algorithm=MD5, username="1065", realm="77.91.132.9", nonce="5152b137", uri="sip:0669197533 at 77.91.132.9", response="446883f3c97a49ea7a9a554a1ba31b6a"
> X-Can-Renegotiate: true
> Contact: <sip:0momhddj at 7cvtd9ihs2e8.invalid;transport=ws;ob>
> Content-Type: application/sdp
> Session-Expires: 90
> Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS
> Supported: timer,ice,outbound
> User-Agent: JsSIP 0.6.26
> Content-Length: 2554
> v=0
> o=- 4785391175048354014 2 IN IP4 127.0.0.1
> s=-
> t=0 0
> a=group:BUNDLE audio video
> a=msid-semantic: WMS cC3clldcCIxZyWm8eHpKdycUakfANCZmV8Br
> m=audio 2313 RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
> c=IN IP4 192.168.88.26
> a=rtcp:2313 IN IP4 192.168.88.26
> a=candidate:97470069 1 udp 2122260223 192.168.88.26 2313 typ host generation 0
> a=candidate:97470069 2 udp 2122260223 192.168.88.26 2313 typ host generation 0
> a=candidate:1263319685 1 tcp 1518280447 192.168.88.26 0 typ host tcptype active generation 0
> a=candidate:1263319685 2 tcp 1518280447 192.168.88.26 0 typ host tcptype active generation 0
> a=ice-ufrag:8nMZ7w8bHdBBoY1a
> a=ice-pwd:CFh1JMRfcT5BoH7aoKQuDgDR
> a=fingerprint:sha-256 6D:5A:B7:7C:8E:1F:2A:F2:8C:DB:49:E4:8D:8F:BE:A0:84:AF:21:E2:A1:D7:DC:29:B1:EE:C3:0C:2C:AF:6C:04
> a=setup:actpass
> a=mid:audio
> a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
> a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
> a=sendrecv
> a=rtcp-mux
> a=rtpmap:111 opus/48000/2
> a=fmtp:111 minptime=10; useinbandfec=1
> a=rtpmap:103 ISAC/16000
> a=rtpmap:104 ISAC/32000
> a=rtpmap:9 G722/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:106 CN/32000
> a=rtpmap:105 CN/16000
> a=rtpmap:13 CN/8000
> a=rtpmap:126 telephone-event/8000
> a=maxptime:60
> a=ssrc:3696151487 cname:jXfPZ33h32Mx9liw
> a=ssrc:3696151487 msid:cC3clldcCIxZyWm8eHpKdycUakfANCZmV8Br 8a2acec3-8511-4d36-9b51-05b8752c2ddd
> a=ssrc:3696151487 mslabel:cC3clldcCIxZyWm8eHpKdycUakfANCZmV8Br
> a=ssrc:3696151487 label:8a2acec3-8511-4d36-9b51-05b8752c2ddd
> m=video 2313 RTP/SAVPF 100 116 117 96
> c=IN IP4 192.168.88.26
> a=rtcp:2313 IN IP4 192.168.88.26
> a=candidate:97470069 1 udp 2122260223 192.168.88.26 2313 typ host generation 0
> a=candidate:97470069 2 udp 2122260223 192.168.88.26 2313 typ host generation 0
> a=candidate:1263319685 1 tcp 1518280447 192.168.88.26 0 typ host tcptype active generation 0
> a=candidate:1263319685 2 tcp 1518280447 192.168.88.26 0 typ host tcptype active generation 0
> a=ice-ufrag:8nMZ7w8bHdBBoY1a
> a=ice-pwd:CFh1JMRfcT5BoH7aoKQuDgDR
> a=fingerprint:sha-256 6D:5A:B7:7C:8E:1F:2A:F2:8C:DB:49:E4:8D:8F:BE:A0:84:AF:21:E2:A1:D7:DC:29:B1:EE:C3:0C:2C:AF:6C:04
> a=setup:actpass
> a=mid:video
> a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
> a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
> a=recvonly
> a=rtcp-mux
> a=rtpmap:100 VP8/90000
> a=rtcp-fb:100 ccm fir
> a=rtcp-fb:100 nack
> a=rtcp-fb:100 nack pli
> a=rtcp-fb:100 goog-remb
> a=rtpmap:116 red/90000
> a=rtpmap:117 ulpfec/90000
> a=rtpmap:96 rtx/90000
> a=fmtp:96 apt=100
> {noformat}
> 2-BUT when i do channel originate sip/GOROD/XXXXX extension 1065 at office
> {noformat}
> -- Executing [1065 at office:1] Dial("SIP/GOROD-00000004", "SIP/1065") in new stack
> == Using SIP RTP CoS mark 5
> [Apr 28 14:07:47] ERROR[4006][C-00000032]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("7cvtd9ihs2e8.invalid", "(null)", ...): Name or service not known
> [Apr 28 14:07:47] WARNING[4006][C-00000032]: chan_sip.c:15869 __set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : '7cvtd9ihs2e8.invalid'
> [Apr 28 14:07:47] ERROR[4006][C-00000032]: netsock2.c:98 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported
> Audio is at 16476
> 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 91.196.158.205:1466:
> INVITE sip:0momhddj at 7cvtd9ihs2e8.invalid;transport=ws SIP/2.0
> Via: SIP/2.0/WS 77.91.132.9:5060;branch=z9hG4bK3f293d79;rport
> Max-Forwards: 70
> From: "asterisk" <sip:asterisk at 77.91.132.9>;tag=as78119d2b
> To: <sip:0momhddj at 7cvtd9ihs2e8.invalid;transport=ws>
> Contact: <sip:asterisk at 77.91.132.9:5060;transport=WS>
> Call-ID: 17a96e0848cdd7d226d3665a36c65c77 at 77.91.132.9:5060
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX 11.15.0
> Date: Tue, 28 Apr 2015 11:07:47 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Content-Type: application/sdp
> Content-Length: 437
> v=0
> o=root 1122885298 1122885298 IN IP4 77.91.132.9
> s=Asterisk PBX 11.15.0
> c=IN IP4 77.91.132.9
> t=0 0
> m=audio 16476 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=ptime:20
> a=connection:new
> a=setup:actpass
> a=fingerprint:SHA-256 CC:82:C8:04:1F:DC:FE:B7:56:27:26:FF:18:CD:BB:71:99:B8:97:F9:81:2B:08:74:72:67:3B:A9:88:5F:00:34
> a=sendrecv
> {noformat}
> thats why i got Failed to set remote offer sdp: Called with SDP without ice-ufrag and ice-pwd
> Waiting for your advice ---thanks
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