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[asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

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jonas.kellens at telen...
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Posted: Tue Aug 09, 2016 3:48 pm Post subject: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get IC

Hello I'm trying for several days now to get ICE support for my Asterisk 11.23 on CentOS 6. My call setup : sipml5_webRTC (nat) --> public Asterisk on 178.18.90.230 --> softphone Zoiper (problem : no audio) Reverse does not work either. (problem : failed get local SDP) I followed this guide : https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5 https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support I researched on the web and found this useful thread : http://forums.digium.com/viewtopic.php?f=1&t=90167 This is no question "what is wrong ?". I know what is wrong : I need ICE support ! So the question here is : how to get ICE support in my Asterisk ? I've compiled asterisk as follow : [root@myserver admin]# yum install uuid-devel libuuid-devel [root@myserver admin]# ./configure --libdir=/usr/lib64 [root@myserver admin]# make menuselect [root@myserver admin]# make && make install In my sip.conf I have : icesupport = yes In my rtp.conf I have : icesupport=yes stunaddr=stun.l.google.com:19302 My SIP peer definition for webRTC client (sipml5) : [770000wrtc] type=peer host=dynamic username=770000wrtc defaultuser=770000wrtc fromuser=770000wrtc secret=987654 disallow=all allow=alaw ;allow=gsm qualify=yes canreinvite=no dtmfmode=rfc2833 amaflags=billing context=testwebrtc nat=force_rport,comedia transport=udp,ws,wss encryption=yes avpf=yes force_avp=yes icesupport=yes directmedia=no dtlsenable=yes dtlsverify=fingerprint dtlscertfile=/etc/asterisk/keys/asterisk.pem dtlscafile=/etc/asterisk/keys/ca.crt dtlssetup=actpass SIP registration works fine : [AugÂ 9 22:12:00]Â Â == WebSocket connection from '178.119.146.190:36940' for protocol 'sip' accepted using version '13' [AugÂ 9 22:12:00]Â Â Â Â -- Registered SIP '770000wrtc' at 178.119.146.190:36940 [AugÂ 9 22:12:00]Â Â Â Â Â Â Â > Saved useragent "IM-client/OMA1.0 sipML5-v1.2016.03.04" for peer 770000wrtc But when I call from my webRTc client (sipml5 website demo) I have no audio. I think this is because there is no ICE support. You can see in de SIP trace below and the RTP trace below that there is no ICE support in Asterisk. [AugÂ 9 22:15:50] <--- SIP read from [url=WS:178.119.146.190:36940]WS:178.119.146.190:36940[/url] ---> [AugÂ 9 22:15:50] INVITE sip:419@178.18.90.230 ([email]sip:419@178.18.90.230[/email]) SIP/2.0 [AugÂ 9 22:15:50] Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKk2KDePVLlTquEfJzIk7LCMdnOHHk4wn1;rport [AugÂ 9 22:15:50] From: "77"<sip:770000wrtc@178.18.90.230> ([email]sip:770000wrtc@178.18.90.230[/email]);tag=sRCvFQq3gUMqkl6TKTcl [AugÂ 9 22:15:50] To: <sip:419@178.18.90.230> ([email]sip:419@178.18.90.230[/email]) [AugÂ 9 22:15:50] Contact: "77"<sips:770000wrtc@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss> ([email]sips:770000wrtc@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss[/email]);+g.oma.sip-im;language="en,fr" [AugÂ 9 22:15:50] Call-ID: 6aa0db27-a37b-69ee-8641-87c5bc444d32 [AugÂ 9 22:15:50] CSeq: 21553 INVITE [AugÂ 9 22:15:50] Content-Type: application/sdp [AugÂ 9 22:15:50] Content-Length: 1815 [AugÂ 9 22:15:50] Max-Forwards: 70 [AugÂ 9 22:15:50] Authorization: Digest username="770000wrtc",realm="178.18.90.230",nonce="1d8fa83d",uri="sip:419@178.18.90.230" ([email]sip:419@178.18.90.230[/email]),response="cd2da8d1cbf0a2795b38b2048a3a3c49",algorithm=MD5 [AugÂ 9 22:15:50] User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04 [AugÂ 9 22:15:50] Organization: Doubango Telecom [AugÂ 9 22:15:50] [AugÂ 9 22:15:50] v=0 [AugÂ 9 22:15:50] o=- 9108976588890881000 2 IN IP4 127.0.0.1 [AugÂ 9 22:15:50] s=Doubango Telecom - chrome [AugÂ 9 22:15:50] t=0 0 [AugÂ 9 22:15:50] a=group:BUNDLE audio [AugÂ 9 22:15:50] a=msid-semantic: WMS BJSlrOtzPj6wzI3QugifY58Oi18zpEbkNsps [AugÂ 9 22:15:50] m=audio 41178 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126 [AugÂ 9 22:15:50] c=IN IP4 178.119.146.190 [AugÂ 9 22:15:50] a=rtcp:42197 IN IP4 178.119.146.190 [AugÂ 9 22:15:50] a=candidate:1668076467 1 udp 2122260223 192.168.1.122 41178 typ host generation 0 [AugÂ 9 22:15:50] a=candidate:1668076467 2 udp 2122260222 192.168.1.122 42197 typ host generation 0 [AugÂ 9 22:15:50] a=candidate:3794064647 1 udp 1686052607 178.119.146.190 41178 typ srflx raddr 192.168.1.122 rport 41178 generation 0 [AugÂ 9 22:15:50] a=candidate:3794064647 2 udp 1686052606 178.119.146.190 42197 typ srflx raddr 192.168.1.122 rport 42197 generation 0 [AugÂ 9 22:15:50] a=candidate:770649923 1 tcp 1518280447 192.168.1.122 0 typ host tcptype active generation 0 [AugÂ 9 22:15:50] a=candidate:770649923 2 tcp 1518280446 192.168.1.122 0 typ host tcptype active generation 0 [AugÂ 9 22:15:50] a=ice-ufrag:cd8nLIL1irEPdLZt [AugÂ 9 22:15:50] a=ice-pwd:97awKXGiAt1TO5jlmb3GMXRy [AugÂ 9 22:15:50] a=fingerprint:sha-256 A2:EF:18:69:AE:9D:D9:90:45:0E:0D:84:5C:A4:AE:59:1C:53:09:11:F2:10:DF:F9:BB:20:E0:9D:6D:ED:BC:13 [AugÂ 9 22:15:50] a=setup:actpass [AugÂ 9 22:15:50] a=mid:audio [AugÂ 9 22:15:50] a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level [AugÂ 9 22:15:50] a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time [AugÂ 9 22:15:50] a=sendrecv [AugÂ 9 22:15:50] a=rtcp-mux [AugÂ 9 22:15:50] a=rtpmap:111 opus/48000/2 [AugÂ 9 22:15:50] a=fmtp:111 minptime=10; useinbandfec=1 [AugÂ 9 22:15:50] a=rtpmap:103 ISAC/16000 [AugÂ 9 22:15:50] a=rtpmap:104 ISAC/32000 [AugÂ 9 22:15:50] a=rtpmap:9 G722/8000 [AugÂ 9 22:15:50] a=rtpmap:0 PCMU/8000 [AugÂ 9 22:15:50] a=rtpmap:8 PCMA/8000 [AugÂ 9 22:15:50] a=rtpmap:106 CN/32000 [AugÂ 9 22:15:50] a=rtpmap:105 CN/16000 [AugÂ 9 22:15:50] a=rtpmap:13 CN/8000 [AugÂ 9 22:15:50] a=rtpmap:126 telephone-event/8000 [AugÂ 9 22:15:50] a=maxptime:60 [AugÂ 9 22:15:50] a=ssrc:1885999682 cname:yLxCKvQLz0YJGRkR [AugÂ 9 22:15:50] a=ssrc:1885999682 msid:BJSlrOtzPj6wzI3QugifY58Oi18zpEbkNsps f0144e6c-86a1-4b08-bf58-4ced92361250 [AugÂ 9 22:15:50] a=ssrc:1885999682 mslabel:BJSlrOtzPj6wzI3QugifY58Oi18zpEbkNsps [AugÂ 9 22:15:50] a=ssrc:1885999682 label:f0144e6c-86a1-4b08-bf58-4ced92361250 [AugÂ 9 22:15:50] <-------------> [AugÂ 9 22:15:50] --- (13 headers 40 lines) --- [AugÂ 9 22:15:50] Using INVITE request as basis request - 6aa0db27-a37b-69ee-8641-87c5bc444d32 [AugÂ 9 22:15:51] WARNING[4349][C-00000001]: res_config_mysql.c:511 realtime_multi_mysql: MySQL RealTime: Failed to query database: Unknown column 'insecure' in 'where clause' [AugÂ 9 22:15:51] WARNING[4349][C-00000001]: res_config_mysql.c:511 realtime_multi_mysql: MySQL RealTime: Failed to query database: Unknown column 'insecure' in 'where clause' [AugÂ 9 22:15:51] Found peer '770000wrtc' for '770000wrtc' from 178.119.146.190:36940 [AugÂ 9 22:15:51]Â Â == Using SIP RTP TOS bits 184 [AugÂ 9 22:15:51]Â Â == Using SIP RTP CoS mark 5 [AugÂ 9 22:15:51] Found RTP audio format 111 [AugÂ 9 22:15:51] Found RTP audio format 103 [AugÂ 9 22:15:51] Found RTP audio format 104 [AugÂ 9 22:15:51] Found RTP audio format 9 [AugÂ 9 22:15:51] Found RTP audio format 0 [AugÂ 9 22:15:51] Found RTP audio format 8 [AugÂ 9 22:15:51] Found RTP audio format 106 [AugÂ 9 22:15:51] Found RTP audio format 105 [AugÂ 9 22:15:51] Found RTP audio format 13 [AugÂ 9 22:15:51] Found RTP audio format 126 [AugÂ 9 22:15:51] Found unknown media description format opus for ID 111 [AugÂ 9 22:15:51] Found unknown media description format ISAC for ID 103 [AugÂ 9 22:15:51] Found unknown media description format ISAC for ID 104 [AugÂ 9 22:15:51] Found audio description format G722 for ID 9 [AugÂ 9 22:15:51] Found audio description format PCMU for ID 0 [AugÂ 9 22:15:51] Found audio description format PCMA for ID 8 [AugÂ 9 22:15:51] Found unknown media description format CN for ID 106 [AugÂ 9 22:15:51] Found unknown media description format CN for ID 105 [AugÂ 9 22:15:51] Found audio description format CN for ID 13 [AugÂ 9 22:15:51] Found audio description format telephone-event for ID 126 [AugÂ 9 22:15:51] Capabilities: us - (alaw), peer - audio=(ulaw\|alaw\|g722)/video=(nothing)/text=(nothing), combined - (alaw) [AugÂ 9 22:15:51] Non-codec capabilities (dtmf): us - 0x1 (telephone-event\|), peer - 0x3 (telephone-event\|CN\|), combined - 0x1 (telephone-event\|) [AugÂ 9 22:15:51] Peer audio RTP is at port 178.119.146.190:41178 [AugÂ 9 22:15:51] Looking for 419 in testwebrtc (domain 178.18.90.230) [AugÂ 9 22:15:51] list_route: hop: <sips:770000wrtc@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss> ([email]sips:770000wrtc@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss[/email]) [AugÂ 9 22:15:51] [AugÂ 9 22:15:51] <--- Transmitting (NAT) to 178.119.146.190:36940 ---> [AugÂ 9 22:15:51] SIP/2.0 100 Trying [AugÂ 9 22:15:51] Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKk2KDePVLlTquEfJzIk7LCMdnOHHk4wn1;received=178.119.146.190;rport=36940 [AugÂ 9 22:15:51] From: "77"<sip:770000wrtc@178.18.90.230> ([email]sip:770000wrtc@178.18.90.230[/email]);tag=sRCvFQq3gUMqkl6TKTcl [AugÂ 9 22:15:51] To: <sip:419@178.18.90.230> ([email]sip:419@178.18.90.230[/email]) [AugÂ 9 22:15:51] Call-ID: 6aa0db27-a37b-69ee-8641-87c5bc444d32 [AugÂ 9 22:15:51] CSeq: 21553 INVITE [AugÂ 9 22:15:51] Server: myPBX [AugÂ 9 22:15:51] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [AugÂ 9 22:15:51] Supported: replaces, timer [AugÂ 9 22:15:51] Contact: <sip:419@178.18.90.230:5060;transport=WS> ([email]sip:419@178.18.90.230:5060;transport=WS[/email]) [AugÂ 9 22:15:51] Content-Length: 0 [AugÂ 9 22:15:51] [AugÂ 9 22:15:51] [AugÂ 9 22:15:51] <------------> [AugÂ 9 22:15:51]Â Â Â Â -- Executing [419@testwebrtc:1] NoOp("SIP/770000wrtc-00000002", "") in new stack [AugÂ 9 22:15:51]Â Â Â Â -- Executing [419@testwebrtc:4] Dial("SIP/770000wrtc-00000002", "SIP/testacc7700905") in new stack [AugÂ 9 22:15:51]Â Â == Using SIP RTP TOS bits 184 [AugÂ 9 22:15:51]Â Â == Using SIP RTP CoS mark 5 [AugÂ 9 22:15:51]Â Â Â Â -- Called SIP/testacc7700905 [AugÂ 9 22:15:51]Â Â Â Â -- SIP/testacc7700905-00000003 is ringing [AugÂ 9 22:15:51] [AugÂ 9 22:15:51] <--- Transmitting (NAT) to 178.119.146.190:36940 ---> [AugÂ 9 22:15:51] SIP/2.0 180 Ringing [AugÂ 9 22:15:51] Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKk2KDePVLlTquEfJzIk7LCMdnOHHk4wn1;received=178.119.146.190;rport=36940 [AugÂ 9 22:15:51] From: "77"<sip:770000wrtc@178.18.90.230> ([email]sip:770000wrtc@178.18.90.230[/email]);tag=sRCvFQq3gUMqkl6TKTcl [AugÂ 9 22:15:51] To: <sip:419@178.18.90.230> ([email]sip:419@178.18.90.230[/email]);tag=as50efde9f [AugÂ 9 22:15:51] Call-ID: 6aa0db27-a37b-69ee-8641-87c5bc444d32 [AugÂ 9 22:15:51] CSeq: 21553 INVITE [AugÂ 9 22:15:51] Server: myPBX [AugÂ 9 22:15:51] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [AugÂ 9 22:15:51] Supported: replaces, timer [AugÂ 9 22:15:51] Contact: <sip:419@178.18.90.230:5060;transport=WS> ([email]sip:419@178.18.90.230:5060;transport=WS[/email]) [AugÂ 9 22:15:51] Content-Length: 0 [AugÂ 9 22:15:51] [AugÂ 9 22:15:51] [AugÂ 9 22:15:51] <------------> [AugÂ 9 22:15:51]Â Â Â Â -- SIP/testacc7700905-00000003 is ringing [AugÂ 9 22:15:52]Â Â Â Â Â Â Â > 0x7fc5dc014060 -- Probation passed - setting RTP source address to 178.119.159.58:44704 [AugÂ 9 22:15:52] NOTICE[4387][C-00000001]: res_rtp_asterisk.c:4476 ast_rtp_read: Unknown RTP codec 95 received from '178.119.159.58:44704' [AugÂ 9 22:15:52]Â Â Â Â -- SIP/testacc7700905-00000003 answered SIP/770000wrtc-00000002 [AugÂ 9 22:15:52] Audio is at 11536 [AugÂ 9 22:15:52] Adding codec 100004 (alaw) to SDP [AugÂ 9 22:15:52] Adding non-codec 0x1 (telephone-event) to SDP [AugÂ 9 22:15:52] [AugÂ 9 22:15:52] <--- Reliably Transmitting (NAT) to 178.119.146.190:36940 ---> [AugÂ 9 22:15:52] SIP/2.0 200 OK [AugÂ 9 22:15:52] Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKk2KDePVLlTquEfJzIk7LCMdnOHHk4wn1;received=178.119.146.190;rport=36940 [AugÂ 9 22:15:52] From: "77"<sip:770000wrtc@178.18.90.230> ([email]sip:770000wrtc@178.18.90.230[/email]);tag=sRCvFQq3gUMqkl6TKTcl [AugÂ 9 22:15:52] To: <sip:419@178.18.90.230> ([email]sip:419@178.18.90.230[/email]);tag=as50efde9f [AugÂ 9 22:15:52] Call-ID: 6aa0db27-a37b-69ee-8641-87c5bc444d32 [AugÂ 9 22:15:52] CSeq: 21553 INVITE [AugÂ 9 22:15:52] Server: myPBX [AugÂ 9 22:15:52] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [AugÂ 9 22:15:52] Supported: replaces, timer [AugÂ 9 22:15:52] Contact: <sip:419@178.18.90.230:5060;transport=WS> ([email]sip:419@178.18.90.230:5060;transport=WS[/email]) [AugÂ 9 22:15:52] Content-Type: application/sdp [AugÂ 9 22:15:52] Content-Length: 387 [AugÂ 9 22:15:52] [AugÂ 9 22:15:52] v=0 [AugÂ 9 22:15:52] o=myPBX 1420513531 1420513531 IN IP4 178.18.90.230 [AugÂ 9 22:15:52] s=myPBX [AugÂ 9 22:15:52] c=IN IP4 178.18.90.230 [AugÂ 9 22:15:52] t=0 0 [AugÂ 9 22:15:52] m=audio 11536 RTP/SAVPF 8 126 [AugÂ 9 22:15:52] a=rtpmap:8 PCMA/8000 [AugÂ 9 22:15:52] a=rtpmap:126 telephone-event/8000 [AugÂ 9 22:15:52] a=fmtp:126 0-16 [AugÂ 9 22:15:52] a=ptime:20 [AugÂ 9 22:15:52] a=connection:new [AugÂ 9 22:15:52] a=setup:active [AugÂ 9 22:15:52] a=fingerprint:SHA-256 DB:10:AC:29:28:3A:55:7A:68:59:57:3C:22:ED:C8:20:4F:79:CC:4E:01:F5:55:10:3D:B4:D2:DD:5B:24:1E:2A [AugÂ 9 22:15:52] a=sendrecv [AugÂ 9 22:15:52] [AugÂ 9 22:15:52] <------------> [AugÂ 9 22:15:52] [AugÂ 9 22:15:52] <--- SIP read from [url=WS:178.119.146.190:36940]WS:178.119.146.190:36940[/url] ---> [AugÂ 9 22:15:52] ACK sip:419@178.18.90.230:5060;transport=WS ([email]sip:419@178.18.90.230:5060;transport=WS[/email]) SIP/2.0 [AugÂ 9 22:15:52] Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKe2fFxswvLSg8fovfxEpP;rport [AugÂ 9 22:15:52] From: "77"<sip:770000wrtc@178.18.90.230> ([email]sip:770000wrtc@178.18.90.230[/email]);tag=sRCvFQq3gUMqkl6TKTcl [AugÂ 9 22:15:52] To: <sip:419@178.18.90.230> ([email]sip:419@178.18.90.230[/email]);tag=as50efde9f [AugÂ 9 22:15:52] Contact: "77"<sips:770000wrtc@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss> ([email]sips:770000wrtc@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss[/email]);+g.oma.sip-im;language="en,fr" [AugÂ 9 22:15:52] Call-ID: 6aa0db27-a37b-69ee-8641-87c5bc444d32 [AugÂ 9 22:15:52] CSeq: 21553 ACK [AugÂ 9 22:15:52] Content-Length: 0 [AugÂ 9 22:15:52] Max-Forwards: 70 [AugÂ 9 22:15:52] Authorization: Digest username="770000wrtc",realm="178.18.90.230",nonce="1d8fa83d",uri="sip:419@178.18.90.230:5060;transport=WS" ([email]sip:419@178.18.90.230:5060;transport=WS[/email]),response="fb65d05b7872c6650836d83535122ef1",algorithm=MD5 [AugÂ 9 22:15:52] User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04 [AugÂ 9 22:15:52] Organization: Doubango Telecom [AugÂ 9 22:15:52] [AugÂ 9 22:15:52] <-------------> [AugÂ 9 22:15:52] --- (12 headers 0 lines) --- [AugÂ 9 22:15:52]Â Â Â Â Â Â Â > 0x7fc5dc014060 -- Probation passed - setting RTP source address to 178.119.159.58:44704 [AugÂ 9 22:15:52]Â Â Â Â Â Â Â > 0x7fc5dc014060 -- Probation passed - setting RTP source address to 178.119.159.58:44704 [AugÂ 9 22:17:08] Sent RTP packet toÂ Â Â Â Â 178.119.146.190:59051 (type 08, seq 028865, ts 2789673216, len 000160) [AugÂ 9 22:17:09] Sent RTP packet toÂ Â Â Â Â 178.119.146.190:59051 (type 08, seq 028866, ts 2789673376, len 000160) [AugÂ 9 22:17:09] Sent RTP packet toÂ Â Â Â Â 178.119.146.190:59051 (type 08, seq 028867, ts 2789673536, len 000160) [AugÂ 9 22:17:09] Sent RTP packet toÂ Â Â Â Â 178.119.146.190:59051 (type 08, seq 028868, ts 2789673696, len 000160) [AugÂ 9 22:17:09] Sent RTP packet toÂ Â Â Â Â 178.119.146.190:59051 (type 08, seq 028869, ts 2789673856, len 000160) [AugÂ 9 22:17:09] Sent RTP packet toÂ Â Â Â Â 178.119.146.190:59051 (type 08, seq 028870, ts 2789674016, len 000160) [AugÂ 9 22:17:09] Sent RTP packet toÂ Â Â Â Â 178.119.146.190:59051 (type 08, seq 028871, ts 2789674176, len 000160) [AugÂ 9 22:17:09] Sent RTP packet toÂ Â Â Â Â 178.119.146.190:59051 (type 08, seq 028872, ts 2789674336, len 000160) [AugÂ 9 22:17:09] Sent RTP packet toÂ Â Â Â Â 178.119.146.190:59051 (type 08, seq 028873, ts 2789674496, len 000160) [AugÂ 9 22:17:09] Sent RTP packet toÂ Â Â Â Â 178.119.146.190:59051 (type 08, seq 028874, ts 2789674656, len 000160) [AugÂ 9 22:17:09] Sent RTP packet toÂ Â Â Â Â 178.119.146.190:59051 (type 08, seq 028875, ts 2789674816, len 000160) [AugÂ 9 22:17:09] Sent RTP packet toÂ Â Â Â Â 178.119.146.190:59051 (type 08, seq 028876, ts 2789674976, len 000160) [AugÂ 9 22:17:09] Sent RTP packet toÂ Â Â Â Â 178.119.146.190:59051 (type 08, seq 028877, ts 2789675136, len 000160) [AugÂ 9 22:17:09] Sent RTP packet toÂ Â Â Â Â 178.119.146.190:59051 (type 08, seq 028878, ts 2789675296, len 000160) [AugÂ 9 22:17:09] Sent RTP packet toÂ Â Â Â Â 178.119.146.190:59051 (type 08, seq 028879, ts 2789675456, len 000160) [AugÂ 9 22:17:09] Sent RTP packet toÂ Â Â Â Â 178.119.146.190:59051 (type 08, seq 028880, ts 2789675616, len 000160) [AugÂ 9 22:17:09] Sent RTP packet toÂ Â Â Â Â 178.119.146.190:59051 (type 08, seq 028881, ts 2789675776, len 000160) [AugÂ 9 22:17:09] Sent RTP packet toÂ Â Â Â Â 178.119.146.190:59051 (type 08, seq 028882, ts 2789675936, len 000160) [AugÂ 9 22:17:09] Sent RTP packet toÂ Â Â Â Â 178.119.146.190:59051 (type 08, seq 028883, ts 2789676096, len 000160) [AugÂ 9 22:17:09] Sent RTP packet toÂ Â Â Â Â 178.119.146.190:59051 (type 08, seq 028884, ts 2789676256, len 000160) [AugÂ 9 22:17:09] Sent RTP packet toÂ Â Â Â Â 178.119.146.190:59051 (type 08, seq 028885, ts 2789676416, len 000160) [AugÂ 9 22:17:09] Sent RTP packet toÂ Â Â Â Â 178.119.146.190:59051 (type 08, seq 028886, ts 2789676576, len 000160) [AugÂ 9 22:17:09] Sent RTP packet toÂ Â Â Â Â 178.119.146.190:59051 (type 08, seq 028887, ts 2789676736, len 000160) [AugÂ 9 22:17:09] Sent RTP packet toÂ Â Â Â Â 178.119.146.190:59051 (type 08, seq 028888, ts 2789676896, len 000160) [AugÂ 9 22:17:09] Sent RTP packet toÂ Â Â Â Â 178.119.146.190:59051 (type 08, seq 028889, ts 2789677056, len 000160) [AugÂ 9 22:17:09] Sent RTP packet toÂ Â Â Â Â 178.119.146.190:59051 (type 08, seq 028890, ts 2789677216, len 000160) [AugÂ 9 22:17:09] Sent RTP packet toÂ Â Â Â Â 178.119.146.190:59051 (type 08, seq 028891, ts 2789677376, len 000160) [AugÂ 9 22:17:09] Sent RTP packet toÂ Â Â Â Â 178.119.146.190:59051 (type 08, seq 028892, ts 2789677536, len 000160) So what am I missing to get ICE support on my Asterisk 11.23.0 ?? Thanks in advance for the feedback. Kind regards.

gmludo at gmail.com
Guest

Posted: Wed Aug 10, 2016 1:52 am Post subject: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get IC

For WebRTC, I recommend you to use Asterisk 13+. Have a nice day. Ludovic Gasc (GMLudo) http://www.gmludo.eu/

jonas.kellens at telen...
Guest

Posted: Wed Aug 10, 2016 5:02 am Post subject: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get IC

On 10-08-16 08:52, Ludovic Gasc wrote:

Quote:

For WebRTC, I recommend you to use Asterisk 13+.
Have a nice day.
Ludovic Gasc (GMLudo)
http://www.gmludo.eu/

Hello

then why is there an option in sip.conf and rtp.conf " icesupport=yes" ??

This is no answer to my question.

So again : what am I missing to get ICE support on my Asterisk 11.23.0 ??

Kind regards.

creslin at digium.com
Guest

Posted: Wed Aug 10, 2016 2:54 pm Post subject: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get IC

I don't see an ice-ufrag or ice-pwd line in the response from
Asterisk, correlating with your suspicion that there is no ICE. Are
you sure that the stun server you're using (the google one) still
works? I haven't tried that server in a while, but I distantly seem
to recall that maybe they shut it down.

Asterisk 13 is a better place to be as well. Asterisk 11 hasn't been
feature updated in a while, and it could be that it could be a number
of patches/fixes behind with regards to webrtc support, particularly
with regards to interoperating with a modern browser version.

Hope that helps,
Matthew Fredrickson

On Wed, Aug 10, 2016 at 5:02 AM, Jonas Kellens <jonas.kellens@telenet.be> wrote:

Quote:

On 10-08-16 08:52, Ludovic Gasc wrote:

For WebRTC, I recommend you to use Asterisk 13+.

Have a nice day.

Ludovic Gasc (GMLudo)
http://www.gmludo.eu/

Hello

then why is there an option in sip.conf and rtp.conf " icesupport=yes" ??

This is no answer to my question.

So again : what am I missing to get ICE support on my Asterisk 11.23.0 ??

Kind regards.

--
_____________________________________________________________________
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New to Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk-users mailing list
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--
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

jonas.kellens at telen...
Guest

Posted: Wed Aug 10, 2016 3:02 pm Post subject: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get IC

Quote:

creslin at digium.com
Guest

Posted: Wed Aug 10, 2016 3:03 pm Post subject: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get IC

My suggestion is to verify and debug against Asterisk 13 first, and
then you can try backing down versions, rather than reverse. WebRTC
is a rapidly moving target, and has required ongoing changes that may
not have made it into older and feature frozen versions of Asterisk.

Matthew Fredrickson

On Wed, Aug 10, 2016 at 3:01 PM, Jonas Kellens <jonas.kellens@telenet.be> wrote:

Quote:

Hello

thank you for your answer.

I don't understand how there are many tutorials and examples on the web
where every time the outcome is a working setup. Very strange I feel now
after my personal experience with Asterisk 11 and webRTC.

You also say Asterisk 13. How about Asterisk 12 then ??

Kind regards.

On 10-08-16 21:53, Matt Fredrickson wrote:

I don't see an ice-ufrag or ice-pwd line in the response from
Asterisk, correlating with your suspicion that there is no ICE. Are
you sure that the stun server you're using (the google one) still
works? I haven't tried that server in a while, but I distantly seem
to recall that maybe they shut it down.

Asterisk 13 is a better place to be as well. Asterisk 11 hasn't been
feature updated in a while, and it could be that it could be a number
of patches/fixes behind with regards to webrtc support, particularly
with regards to interoperating with a modern browser version.

Hope that helps,
Matthew Fredrickson

On Wed, Aug 10, 2016 at 5:02 AM, Jonas Kellens <jonas.kellens@telenet.be>
wrote:

On 10-08-16 08:52, Ludovic Gasc wrote:

For WebRTC, I recommend you to use Asterisk 13+.

Have a nice day.

Ludovic Gasc (GMLudo)
http://www.gmludo.eu/

Hello

then why is there an option in sip.conf and rtp.conf " icesupport=yes" ??

This is no answer to my question.

So again : what am I missing to get ICE support on my Asterisk 11.23.0 ??

Kind regards.

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

jonas.kellens at telen...
Guest

Posted: Thu Aug 11, 2016 9:10 am Post subject: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get IC

Hello

Using Asterisk 12.8.2.

I now have the "via ICE" messages in the RTP debug (see below).

If you look in the SIP debug (see below), you also now see the
"ice-ufrag" and "ice-pwd" in the 200 OK SIP-message from Asterisk to the
webRTC client.

But still no audio ! None at all ! In both directions.

You can see in the SIP debug that the IP-address in de SDP-body is
correctly set for sending audio. So I don't think it is a NAT/ICE problem.

Can anyone tell me then what is left that could be causing the
'no-audio' problem ??

SIP debug :

[Aug 11 15:53:47] <--- SIP read from WS:178.119.146.190:60191 --->
[Aug 11 15:53:47] INVITE sip:419@178.18.90.230 SIP/2.0
[Aug 11 15:53:47] Via: SIP/2.0/WSS
df7jal23ls0d.invalid;branch=z9hG4bKSqKu6K3uxr3dOFdU5WAtPM5tKKA5yzAq;rport
[Aug 11 15:53:47] From:
<sip:770000wrtc@178.18.90.230>;tag=SGUVL1LMdvxQrUfxprZJ
[Aug 11 15:53:47] To: <sip:419@178.18.90.230>
[Aug 11 15:53:47] Contact:
<sips:770000wrtc@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss>;+g.oma.sip-im;language="en,fr"
[Aug 11 15:53:47] Call-ID: 47ca4cc9-9dce-4449-d58f-e069a67061ec
[Aug 11 15:53:47] CSeq: 58874 INVITE
[Aug 11 15:53:47] Content-Type: application/sdp
[Aug 11 15:53:47] Content-Length: 2301
[Aug 11 15:53:47] Max-Forwards: 70
[Aug 11 15:53:47] Authorization: Digest
username="770000wrtc",realm="178.18.90.230",nonce="5d5c700b",uri="sip:419@178.18.90.230",response="ca118222a4674b4c6dcc19dd95e00c15",algorithm=MD5
[Aug 11 15:53:47] User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
[Aug 11 15:53:47] Organization: Doubango Telecom
[Aug 11 15:53:47]
[Aug 11 15:53:47] v=0
[Aug 11 15:53:47] o=- 5876454736929512000 2 IN IP4 127.0.0.1
[Aug 11 15:53:47] s=Doubango Telecom - chrome
[Aug 11 15:53:47] t=0 0
[Aug 11 15:53:47] a=group:BUNDLE audio
[Aug 11 15:53:47] a=msid-semantic: WMS kBwsfCPizGNiVjZS23dGoTNcUubDMMhxXrka
[Aug 11 15:53:47] m=audio 63897 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106
105 13 126
[Aug 11 15:53:47] c=IN IP4 178.119.146.190
[Aug 11 15:53:47] a=rtcp:63899 IN IP4 178.119.146.190
[Aug 11 15:53:47] a=candidate:2999745851 1 udp 2122260223 192.168.56.1
63896 typ host generation 0 network-id 1
[Aug 11 15:53:47] a=candidate:3378846520 1 udp 2122194687 192.168.1.120
63897 typ host generation 0 network-id 2
[Aug 11 15:53:47] a=candidate:2999745851 2 udp 2122260222 192.168.56.1
63898 typ host generation 0 network-id 1
[Aug 11 15:53:47] a=candidate:3378846520 2 udp 2122194686 192.168.1.120
63899 typ host generation 0 network-id 2
[Aug 11 15:53:47] a=candidate:1210916236 1 udp 1685987071
178.119.146.190 63897 typ srflx raddr 192.168.1.120 rport 63897
generation 0 network-id 2
[Aug 11 15:53:47] a=candidate:1210916236 2 udp 1685987070
178.119.146.190 63899 typ srflx raddr 192.168.1.120 rport 63899
generation 0 network-id 2
[Aug 11 15:53:47] a=candidate:4233069003 1 tcp 1518280447 192.168.56.1 9
typ host tcptype active generation 0 network-id 1
[Aug 11 15:53:47] a=candidate:2280056776 1 tcp 1518214911 192.168.1.120
9 typ host tcptype active generation 0 network-id 2
[Aug 11 15:53:47] a=candidate:4233069003 2 tcp 1518280446 192.168.56.1 9
typ host tcptype active generation 0 network-id 1
[Aug 11 15:53:47] a=candidate:2280056776 2 tcp 1518214910 192.168.1.120
9 typ host tcptype active generation 0 network-id 2
[Aug 11 15:53:47] a=ice-ufrag:TxJQpv1i5O04Q+Kw
[Aug 11 15:53:47] a=ice-pwd:LvfUjrDPbY/np215T3+6Sy03
[Aug 11 15:53:47] a=fingerprint:sha-256
EF:A4:78:E4:C1:33:5A:F5:36:6B:C5:DF:C7:D9:10:44:FD:96:5D:88:79:AB:8C:A0:E2:71:66:DA:6D:2C:30:84
[Aug 11 15:53:47] a=setup:actpass
[Aug 11 15:53:47] a=mid:audio
[Aug 11 15:53:47] a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
[Aug 11 15:53:47] a=extmap:3
http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
[Aug 11 15:53:47] a=sendrecv
[Aug 11 15:53:47] a=rtcp-mux
[Aug 11 15:53:47] a=rtpmap:111 opus/48000/2
[Aug 11 15:53:47] a=rtcp-fb:111 transport-cc
[Aug 11 15:53:47] a=fmtp:111 minptime=10;useinbandfec=1
[Aug 11 15:53:47] a=rtpmap:103 ISAC/16000
[Aug 11 15:53:47] a=rtpmap:104 ISAC/32000
[Aug 11 15:53:47] a=rtpmap:9 G722/8000
[Aug 11 15:53:47] a=rtpmap:0 PCMU/8000
[Aug 11 15:53:47] a=rtpmap:8 PCMA/8000
[Aug 11 15:53:47] a=rtpmap:106 CN/32000
[Aug 11 15:53:47] a=rtpmap:105 CN/16000
[Aug 11 15:53:47] a=rtpmap:13 CN/8000
[Aug 11 15:53:47] a=rtpmap:126 telephone-event/8000
[Aug 11 15:53:47] a=ssrc:54412034 cname:H2asKiJklFa9L3Xw
[Aug 11 15:53:47] a=ssrc:54412034
msid:kBwsfCPizGNiVjZS23dGoTNcUubDMMhxXrka
f25030f2-3e48-4180-aea4-4edec3e67410
[Aug 11 15:53:47] a=ssrc:54412034
mslabel:kBwsfCPizGNiVjZS23dGoTNcUubDMMhxXrka
[Aug 11 15:53:47] a=ssrc:54412034 label:f25030f2-3e48-4180-aea4-4edec3e67410
[Aug 11 15:53:47] <------------->
[Aug 11 15:53:47] --- (13 headers 44 lines) ---
[Aug 11 15:53:47] Using INVITE request as basis request -
47ca4cc9-9dce-4449-d58f-e069a67061ec
[Aug 11 15:53:47] Found peer '770000wrtc' for '770000wrtc' from
178.119.146.190:60191
[Aug 11 15:53:47] == Using SIP RTP TOS bits 184
[Aug 11 15:53:47] == Using SIP RTP CoS mark 5
[Aug 11 15:53:47] Found RTP audio format 111
[Aug 11 15:53:47] Found RTP audio format 103
[Aug 11 15:53:47] Found RTP audio format 104
[Aug 11 15:53:47] Found RTP audio format 9
[Aug 11 15:53:47] Found RTP audio format 0
[Aug 11 15:53:47] Found RTP audio format 8
[Aug 11 15:53:47] Found RTP audio format 106
[Aug 11 15:53:47] Found RTP audio format 105
[Aug 11 15:53:47] Found RTP audio format 13
[Aug 11 15:53:47] Found RTP audio format 126
[Aug 11 15:53:47] Found audio description format opus for ID 111
[Aug 11 15:53:47] Found unknown media description format ISAC for ID 103
[Aug 11 15:53:47] Found unknown media description format ISAC for ID 104
[Aug 11 15:53:47] Found audio description format G722 for ID 9
[Aug 11 15:53:47] Found audio description format PCMU for ID 0
[Aug 11 15:53:47] Found audio description format PCMA for ID 8
[Aug 11 15:53:47] Found unknown media description format CN for ID 106
[Aug 11 15:53:47] Found unknown media description format CN for ID 105
[Aug 11 15:53:47] Found audio description format CN for ID 13
[Aug 11 15:53:47] Found audio description format telephone-event for ID 126
[Aug 11 15:53:47] Capabilities: us - (alaw), peer -
audio=(ulaw|alaw|g722|opus)/video=(nothing)/text=(nothing), combined -
(alaw)
[Aug 11 15:53:47] Non-codec capabilities (dtmf): us - 0x1
(telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1
(telephone-event|)
[Aug 11 15:53:47] Peer audio RTP is at port 178.119.146.190:63897
[Aug 11 15:53:47] Looking for 419 in testwebrtc (domain 178.18.90.230)
[Aug 11 15:53:47] list_route: route/path hop:
<sips:770000wrtc@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss>
[Aug 11 15:53:47]
[Aug 11 15:53:47] <--- Transmitting (NAT) to 178.119.146.190:60191 --->
[Aug 11 15:53:47] SIP/2.0 100 Trying
[Aug 11 15:53:47] Via: SIP/2.0/WSS
df7jal23ls0d.invalid;branch=z9hG4bKSqKu6K3uxr3dOFdU5WAtPM5tKKA5yzAq;received=178.119.146.190;rport=60191
[Aug 11 15:53:47] From:
<sip:770000wrtc@178.18.90.230>;tag=SGUVL1LMdvxQrUfxprZJ
[Aug 11 15:53:47] To: <sip:419@178.18.90.230>
[Aug 11 15:53:47] Call-ID: 47ca4cc9-9dce-4449-d58f-e069a67061ec
[Aug 11 15:53:47] CSeq: 58874 INVITE
[Aug 11 15:53:47] Server: myPBX
[Aug 11 15:53:47] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Aug 11 15:53:47] Supported: replaces
[Aug 11 15:53:47] Contact: <sip:419@178.18.90.230:5060;transport=WS>
[Aug 11 15:53:47] Content-Length: 0
[Aug 11 15:53:47]
[Aug 11 15:53:47]
[Aug 11 15:53:47] <------------>
[Aug 11 15:53:47] == Using SIP RTP TOS bits 184
[Aug 11 15:53:47] == Using SIP RTP CoS mark 5
[Aug 11 15:53:47] -- Called SIP/testacc7700905
[Aug 11 15:53:48] -- SIP/testacc7700905-00000001 is ringing
[Aug 11 15:53:48]
[Aug 11 15:53:48] <--- Transmitting (NAT) to 178.119.146.190:60191 --->
[Aug 11 15:53:48] SIP/2.0 180 Ringing
[Aug 11 15:53:48] Via: SIP/2.0/WSS
df7jal23ls0d.invalid;branch=z9hG4bKSqKu6K3uxr3dOFdU5WAtPM5tKKA5yzAq;received=178.119.146.190;rport=60191
[Aug 11 15:53:48] From:
<sip:770000wrtc@178.18.90.230>;tag=SGUVL1LMdvxQrUfxprZJ
[Aug 11 15:53:48] To: <sip:419@178.18.90.230>;tag=as6a3f0437
[Aug 11 15:53:48] Call-ID: 47ca4cc9-9dce-4449-d58f-e069a67061ec
[Aug 11 15:53:48] CSeq: 58874 INVITE
[Aug 11 15:53:48] Server: myPBX
[Aug 11 15:53:48] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Aug 11 15:53:48] Supported: replaces
[Aug 11 15:53:48] Contact: <sip:419@178.18.90.230:5060;transport=WS>
[Aug 11 15:53:48] Content-Length: 0
[Aug 11 15:53:48]
[Aug 11 15:53:48]
[Aug 11 15:53:48] <------------>
[Aug 11 15:53:50] > 0x7f2d8c018ee0 -- Probation passed - setting
RTP source address to 178.119.146.190:58814
[Aug 11 15:53:50] NOTICE[8910][C-00000000]: res_rtp_asterisk.c:4467
ast_rtp_read: Unknown RTP codec 95 received from '178.119.146.190:58814'
[Aug 11 15:53:50] -- SIP/testacc7700905-00000001 answered
SIP/770000wrtc-00000000
[Aug 11 15:53:50] Audio is at 11780
[Aug 11 15:53:50] Adding codec 100004 (alaw) to SDP
[Aug 11 15:53:50] Adding non-codec 0x1 (telephone-event) to SDP
[Aug 11 15:53:50]
[Aug 11 15:53:50] <--- Reliably Transmitting (NAT) to
178.119.146.190:60191 --->
[Aug 11 15:53:50] SIP/2.0 200 OK
[Aug 11 15:53:50] Via: SIP/2.0/WSS
df7jal23ls0d.invalid;branch=z9hG4bKSqKu6K3uxr3dOFdU5WAtPM5tKKA5yzAq;received=178.119.146.190;rport=60191
[Aug 11 15:53:50] From:
<sip:770000wrtc@178.18.90.230>;tag=SGUVL1LMdvxQrUfxprZJ
[Aug 11 15:53:50] To: <sip:419@178.18.90.230>;tag=as6a3f0437
[Aug 11 15:53:50] Call-ID: 47ca4cc9-9dce-4449-d58f-e069a67061ec
[Aug 11 15:53:50] CSeq: 58874 INVITE
[Aug 11 15:53:50] Server: myPBX
[Aug 11 15:53:50] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Aug 11 15:53:50] Supported: replaces
[Aug 11 15:53:50] Contact: <sip:419@178.18.90.230:5060;transport=WS>
[Aug 11 15:53:50] Content-Type: application/sdp
[Aug 11 15:53:50] Content-Length: 969
[Aug 11 15:53:50]
[Aug 11 15:53:50] v=0
[Aug 11 15:53:50] o=myPBX 794545698 794545698 IN IP4 178.18.90.230
[Aug 11 15:53:50] s=myPBX
[Aug 11 15:53:50] c=IN IP4 178.18.90.230
[Aug 11 15:53:50] t=0 0
[Aug 11 15:53:50] m=audio 11780 UDP/TLS/RTP/SAVPF 8 126
[Aug 11 15:53:50] a=rtpmap:8 PCMA/8000
[Aug 11 15:53:50] a=rtpmap:126 telephone-event/8000
[Aug 11 15:53:50] a=fmtp:126 0-16
[Aug 11 15:53:50] a=ptime:20
[Aug 11 15:53:50] a=maxptime:150
[Aug 11 15:53:50] a=ice-ufrag:58a5f9de0d48369c30dba971059275db
[Aug 11 15:53:50] a=ice-pwd:0f085841667af68d2ebc1a055613d53e
[Aug 11 15:53:50] a=candidate:Hb21259ee 1 UDP 2130706431 178.18.90.230
11780 typ host
[Aug 11 15:53:50] a=candidate:Ha0a0101 1 UDP 2130706431 10.10.1.1 11780
typ host
[Aug 11 15:53:50] a=candidate:Sb21259ee 1 UDP 1694498815 178.18.90.230
11780 typ srflx raddr 178.18.90.230 rport 11780
[Aug 11 15:53:50] a=candidate:Hb21259ee 2 UDP 2130706430 178.18.90.230
11781 typ host
[Aug 11 15:53:50] a=candidate:Ha0a0101 2 UDP 2130706430 10.10.1.1 11781
typ host
[Aug 11 15:53:50] a=candidate:Sb21259ee 2 UDP 1694498814 178.18.90.230
11781 typ srflx raddr 178.18.90.230 rport 11781
[Aug 11 15:53:50] a=connection:new
[Aug 11 15:53:50] a=setup:active
[Aug 11 15:53:50] a=fingerprint:SHA-256
DB:10:AC:29:28:3A:55:7A:68:59:57:3C:22:ED:C8:20:4F:79:CC:4E:01:F5:55:10:3D:B4:D2:DD:5B:24:1E:2A
[Aug 11 15:53:50] a=sendrecv
[Aug 11 15:53:50]
[Aug 11 15:53:50] <------------>
[Aug 11 15:53:50] -- Channel SIP/770000wrtc-00000000 joined
'simple_bridge' basic-bridge <eca525df-711a-4f04-a575-71ce917b49e4>
[Aug 11 15:53:50] -- Channel SIP/testacc7700905-00000001 joined
'simple_bridge' basic-bridge <eca525df-711a-4f04-a575-71ce917b49e4>
[Aug 11 15:53:50]
[Aug 11 15:53:50] <--- SIP read from WS:178.119.146.190:60191 --->
[Aug 11 15:53:50] ACK sip:419@178.18.90.230:5060;transport=WS SIP/2.0
[Aug 11 15:53:50] Via: SIP/2.0/WSS
df7jal23ls0d.invalid;branch=z9hG4bKosvPUfE7SGqs3pZo6muw;rport
[Aug 11 15:53:50] From:
<sip:770000wrtc@178.18.90.230>;tag=SGUVL1LMdvxQrUfxprZJ
[Aug 11 15:53:50] To: <sip:419@178.18.90.230>;tag=as6a3f0437
[Aug 11 15:53:50] Contact:
<sips:770000wrtc@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss>;+g.oma.sip-im;language="en,fr"
[Aug 11 15:53:50] Call-ID: 47ca4cc9-9dce-4449-d58f-e069a67061ec
[Aug 11 15:53:50] CSeq: 58874 ACK
[Aug 11 15:53:50] Content-Length: 0
[Aug 11 15:53:50] Max-Forwards: 70
[Aug 11 15:53:50] Authorization: Digest
username="770000wrtc",realm="178.18.90.230",nonce="5d5c700b",uri="sip:419@178.18.90.230:5060;transport=WS",response="426b1c5b355ea70b9d23e3f5af161681",algorithm=MD5
[Aug 11 15:53:50] User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
[Aug 11 15:53:50] Organization: Doubango Telecom

RTP debug :

RTP Debugging Enabled for address: 178.119.146.190:0
[Aug 11 16:01:20] Got RTP packet from 178.119.146.190:58814 (type
08, seq 014114, ts 3292374327, len 000160)
[Aug 11 16:01:20] Sent RTP packet to 178.119.146.190:50760 (via
ICE) (type 08, seq 033787, ts 3292374320, len 000160)
[Aug 11 16:01:20] Got RTP packet from 178.119.146.190:58814 (type
08, seq 014115, ts 3292374487, len 000160)
[Aug 11 16:01:20] Sent RTP packet to 178.119.146.190:50760 (via
ICE) (type 08, seq 033788, ts 3292374480, len 000160)
[Aug 11 16:01:20] Got RTP packet from 178.119.146.190:58814 (type
08, seq 014116, ts 3292374647, len 000160)
[Aug 11 16:01:20] Sent RTP packet to 178.119.146.190:50760 (via
ICE) (type 08, seq 033789, ts 3292374640, len 000160)
[Aug 11 16:01:20] Got RTP packet from 178.119.146.190:58814 (type
08, seq 014117, ts 3292374807, len 000160)
[Aug 11 16:01:20] Sent RTP packet to 178.119.146.190:50760 (via
ICE) (type 08, seq 033790, ts 3292374800, len 000160)
[Aug 11 16:01:20] Got RTP packet from 178.119.146.190:58814 (type
08, seq 014118, ts 3292374967, len 000160)
[Aug 11 16:01:20] Sent RTP packet to 178.119.146.190:50760 (via
ICE) (type 08, seq 033791, ts 3292374960, len 000160)
[Aug 11 16:01:20] Got RTP packet from 178.119.146.190:58814 (type
08, seq 014119, ts 3292375127, len 000160)
[Aug 11 16:01:20] Sent RTP packet to 178.119.146.190:50760 (via
ICE) (type 08, seq 033792, ts 3292375120, len 000160)
[Aug 11 16:01:20] Got RTP packet from 178.119.146.190:58814 (type
08, seq 014120, ts 3292375287, len 000160)
[Aug 11 16:01:20] Sent RTP packet to 178.119.146.190:50760 (via
ICE) (type 08, seq 033793, ts 3292375280, len 000160)
[Aug 11 16:01:20] Got RTP packet from 178.119.146.190:58814 (type
08, seq 014121, ts 3292375447, len 000160)
[Aug 11 16:01:20] Sent RTP packet to 178.119.146.190:50760 (via
ICE) (type 08, seq 033794, ts 3292375440, len 000160)
[Aug 11 16:01:20] Got RTP packet from 178.119.146.190:58814 (type
08, seq 014122, ts 3292375607, len 000160)
[Aug 11 16:01:20] Sent RTP packet to 178.119.146.190:50760 (via
ICE) (type 08, seq 033795, ts 3292375600, len 000160)
[Aug 11 16:01:20] Got RTP packet from 178.119.146.190:58814 (type
08, seq 014123, ts 3292375767, len 000160)
[Aug 11 16:01:20] Sent RTP packet to 178.119.146.190:50760 (via
ICE) (type 08, seq 033796, ts 3292375760, len 000160)
[Aug 11 16:01:20] Got RTP packet from 178.119.146.190:58814 (type
08, seq 014124, ts 3292375927, len 000160)
[Aug 11 16:01:20] Sent RTP packet to 178.119.146.190:50760 (via
ICE) (type 08, seq 033797, ts 3292375920, len 000160)
[Aug 11 16:01:20] Got RTP packet from 178.119.146.190:58814 (type
08, seq 014125, ts 3292376087, len 000160)
[Aug 11 16:01:20] Sent RTP packet to 178.119.146.190:50760 (via
ICE) (type 08, seq 033798, ts 3292376080, len 000160)
[Aug 11 16:01:20] Got RTP packet from 178.119.146.190:58814 (type
08, seq 014126, ts 3292376247, len 000160)
[Aug 11 16:01:20] Sent RTP packet to 178.119.146.190:50760 (via
ICE) (type 08, seq 033799, ts 3292376240, len 000160)
[Aug 11 16:01:20] Got RTP packet from 178.119.146.190:58814 (type
08, seq 014127, ts 3292376407, len 000160)
[Aug 11 16:01:20] Sent RTP packet to 178.119.146.190:50760 (via
ICE) (type 08, seq 033800, ts 3292376400, len 000160)
[Aug 11 16:01:20] Got RTP packet from 178.119.146.190:58814 (type
08, seq 014128, ts 3292376567, len 000160)

On 10-08-16 22:03, Matt Fredrickson wrote:

Quote:

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk-users mailing list
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lardconcepts at gmail.com
Guest

Posted: Thu Aug 11, 2016 9:28 am Post subject: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get IC

I'm genuinely fascinated why you are insisting on using a version of Asterisk almost 3 years old, for which EOL support ended last year.

Is there any particular reason you cannot or will not use the current version as others have suggested?

Also, I see you are using Doubango and WebRTC, but in the logs, I see WS and WSS.

You NEED to be using 100% WSS otherwise you've not got a hope in hell of anything working with WEBRTC.
Check the console of the web browser you are trying to make the call from (CTRL-SHIFT-I in Chrome on Windows, for example).

Also, you'll need to be using valid certificates - self-signed certificates won't work for any current implementation of WebRTC that I know of, certainly not if anything involves current versions of Chrome or Firefox. That said, LetsEncrypt certs work fine for this, so no need to spend out on one.

Switch to Asterisk 13.10 and save yourself a whole lotta headache.

On 11 August 2016 at 15:09, Jonas Kellens <jonas.kellens@telenet.be (jonas.kellens@telenet.be)> wrote:

Quote:

Hello

Using Asterisk 12.8.2.

Quote:

On 10-08-16 22:03, Matt Fredrickson wrote:

Quote:

My suggestion is to verify and debug against Asterisk 13 first, and
then you can try backing down versions, rather than reverse.Â WebRTC
is a rapidly moving target, and has required ongoing changes that may
not have made it into older and feature frozen versions of Asterisk.

jonas.kellens at telen...
Guest

Posted: Thu Aug 11, 2016 9:40 am Post subject: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get IC

My main reason not to upgrade to Ast 13 is because I'm afraid of losing functionality as there are certain functions deprecated/replaced. This can also cause headache Smile

I will do so if there is no other option.

But still, I don't see why Ast 13 would differ so much in this case ? If ICE and NAT is working (not causing problems) why should Ast 13 bring me audio and Ast 12 don't ??

I indeed use SIPML5 demo as quick test-case. So do many tutorials on the web.

Self-signed certificates should be OK as long as they are imported in the browser. Never knew this could cause audio problems ?

Kind regards.

On 11-08-16 16:25, Jonathan H wrote:

Quote:

Hello

Using Asterisk 12.8.2.

Quote:

On 10-08-16 22:03, Matt Fredrickson wrote:

Quote:

creslin at digium.com
Guest

Posted: Thu Aug 11, 2016 11:04 am Post subject: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get IC

On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens <jonas.kellens@telenet.be> wrote:

Quote:

My main reason not to upgrade to Ast 13 is because I'm afraid of losing
functionality as there are certain functions deprecated/replaced. This can
also cause headache Smile

If you want to minimize grief, start with 13 - WebRTC has been a
moving target for the last 5 years, it is not an old, mature standard
like ISDN or SIP. If you find interop problems in an older version of
Asterisk with WebRTC, it's likely that it has been fixed in 13, and if
it hasn't the most likely place to obtain the fix will be in 13.

After you get the WebRTC part working, then you can move back the
versions of Asterisk you're using to see if it still works.

As far as ICE not working goes, if the browser you're talking to is
not on the same network as the Asterisk server, it's *possible* you
might need a true TURN server as well, instead of just an ICE server.

Matthew Fredrickson

Quote:

On 11-08-16 16:25, Jonathan H wrote:

I'm genuinely fascinated why you are insisting on using a version of
Asterisk almost 3 years old, for which EOL support ended last year.

Is there any particular reason you cannot or will not use the current
version as others have suggested?

Also, I see you are using Doubango and WebRTC, but in the logs, I see WS and
WSS.

You NEED to be using 100% WSS otherwise you've not got a hope in hell of
anything working with WEBRTC.
Check the console of the web browser you are trying to make the call from
(CTRL-SHIFT-I in Chrome on Windows, for example).

Also, you'll need to be using valid certificates - self-signed certificates
won't work for any current implementation of WebRTC that I know of,
certainly not if anything involves current versions of Chrome or Firefox.
That said, LetsEncrypt certs work fine for this, so no need to spend out on
one.

Switch to Asterisk 13.10 and save yourself a whole lotta headache.

On 11 August 2016 at 15:09, Jonas Kellens <jonas.kellens@telenet.be> wrote:

Quote:

Hello

Using Asterisk 12.8.2.

Quote:

On 10-08-16 22:03, Matt Fredrickson wrote:

Quote:

mjordan at digium.com
Guest

Posted: Thu Aug 11, 2016 1:44 pm Post subject: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get IC

On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens <jonas.kellens@telenet.be> wrote:

Quote:

My main reason not to upgrade to Ast 13 is because I'm afraid of losing
functionality as there are certain functions deprecated/replaced. This can
also cause headache Smile

What in particular?

Any longer, Asterisk is *very* conservative with functionality that is
removed. Given that Asterisk 13 is simply the evolution and refinement
of the architecture introduced in Asterisk 12, I would not expect
there to be any major differences moving from 12 to 13.

Quote:

Asterisk 13 has a lot more bug fixes than Asterisk 12. Asterisk 12 is
no longer actively supported.

Supported timelines for versions are available on the wiki:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

--
Matthew Jordan
Digium, Inc. | CTO
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

jonas.kellens at telen...
Guest

Posted: Thu Aug 11, 2016 4:01 pm Post subject: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get IC

On 11-08-16 18:03, Matt Fredrickson wrote:

Quote:

On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens <jonas.kellens@telenet.be> wrote:

Quote:

My main reason not to upgrade to Ast 13 is because I'm afraid of losing
functionality as there are certain functions deprecated/replaced. This can
also cause headache Smile

Matthew

when I set the following in rtp.conf :

turnaddr=192.158.29.39:3478?transport=udp
turnusername=28224511:1379330808
turnpassword=JZEOEt2V3Qb0y27GRntt2u2PAYA

then Asterisk 12 gets really slow and sometimes unresponsive. Calls
result in 480 request timeout (possibly due to the freeze of Asterisk).

So this is also no solution.

Can not even test if it brings me some audio in my webRTC calls.

(putting the above lines back in comment resolves the issue of Asterisk
freeze. This is all EXTREMELY BUGGY !)

Asterisk 13 here I come (with very high expectations).

Kind regards.

--
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satskiy.a at gmail.com
Guest

Posted: Fri Aug 12, 2016 2:26 am Post subject: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get IC

Try delete nat from 770000wrtc settings ice should do the same
On Aug 11, 2016 10:00 PM, "Jonas Kellens" <jonas.kellens@telenet.be (jonas.kellens@telenet.be)> wrote:

Quote:

On 11-08-16 18:03, Matt Fredrickson wrote:

Quote:

On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens <jonas.kellens@telenet.be (jonas.kellens@telenet.be)> wrote:

Quote:

My main reason not to upgrade to Ast 13 is because I'm afraid of losing
functionality as there are certain functions deprecated/replaced. This can
also cause headache Smile

If you want to minimize grief, start with 13 - WebRTC has been a
moving target for the last 5 years, it is not an old, mature standard
like ISDN or SIP.Â If you find interop problems in an older version of
Asterisk with WebRTC, it's likely that it has been fixed in 13, and if
it hasn't the most likely place to obtain the fix will be in 13.

After you get the WebRTC part working, then you can move back the
versions of Asterisk you're using to see if it still works.

As far as ICE not working goes, if the browser you're talking to is
not on the same network as the Asterisk server, it's *possible* you
might need a true TURN server as well, instead of just an ICE server.

Matthew Fredrickson

Matthew

when I set the following in rtp.conf :

turnaddr=192.158.29.39:3478?transport=udp
turnusername=28224511:1379330808
turnpassword=JZEOEt2V3Qb0y27GRntt2u2PAYA

then Asterisk 12 gets really slow and sometimes unresponsive. Calls result in 480 request timeout (possibly due to the freeze of Asterisk).

So this is also no solution.

Can not even test if it brings me some audio in my webRTC calls.

(putting the above lines back in comment resolves the issue of Asterisk freeze. This is all EXTREMELY BUGGY !)

Asterisk 13 here I come (with very high expectations).

Kind regards.

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
Â Â Â Â Â Â Â http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Â http://lists.digium.com/mailman/listinfo/asterisk-users

jonas.kellens at telen...
Guest

Posted: Fri Aug 12, 2016 8:02 am Post subject: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get IC

Hello

setting "nat=no" or omitting "nat=" in peer definition does not help either. Still no audio.

Why do you think this is a NAT issue ? IP and port information in SDP-body is correct.

Kind regards.

On 12-08-16 09:25, ÐÐ½Ñ‚Ð¾Ð½ Ð¡Ð°Ñ†ÐºÐ¸Ð¹ wrote:

Quote:

Try delete nat from 770000wrtc settings ice should do the same
On Aug 11, 2016 10:00 PM, "Jonas Kellens" <jonas.kellens@telenet.be (jonas.kellens@telenet.be)> wrote:

Quote:

On 11-08-16 18:03, Matt Fredrickson wrote:

Quote:

On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens <jonas.kellens@telenet.be (jonas.kellens@telenet.be)> wrote:

Quote:

My main reason not to upgrade to Ast 13 is because I'm afraid of losing
functionality as there are certain functions deprecated/replaced. This can
also cause headache Smile

Matthew

when I set the following in rtp.conf :

turnaddr=192.158.29.39:3478?transport=udp
turnusername=28224511:1379330808
turnpassword=JZEOEt2V3Qb0y27GRntt2u2PAYA

then Asterisk 12 gets really slow and sometimes unresponsive. Calls result in 480 request timeout (possibly due to the freeze of Asterisk).

So this is also no solution.

Can not even test if it brings me some audio in my webRTC calls.

(putting the above lines back in comment resolves the issue of Asterisk freeze. This is all EXTREMELY BUGGY !)

Asterisk 13 here I come (with very high expectations).

Kind regards.

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
Â Â Â Â Â Â Â http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Â http://lists.digium.com/mailman/listinfo/asterisk-users

jonas.kellens at telen...
Guest

Posted: Fri Aug 12, 2016 9:23 am Post subject: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get IC

Question : I noticed I received an error when installing pjproject --with-external-srtp

I do not seems to have the srtp capability.
(However I can easily install with "yum install libsrtp-devel")

Can this have anything to do with the no-audio-problems that I'm having ??

Kind regards.

On 12-08-16 15:02, Jonas Kellens wrote:

Quote:

Try delete nat from 770000wrtc settings ice should do the same
On Aug 11, 2016 10:00 PM, "Jonas Kellens" <jonas.kellens@telenet.be (jonas.kellens@telenet.be)> wrote:

Quote:

On 11-08-16 18:03, Matt Fredrickson wrote:

Quote:

On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens <jonas.kellens@telenet.be (jonas.kellens@telenet.be)> wrote:

Quote:

My main reason not to upgrade to Ast 13 is because I'm afraid of losing
functionality as there are certain functions deprecated/replaced. This can
also cause headache Smile

Matthew

when I set the following in rtp.conf :

turnaddr=192.158.29.39:3478?transport=udp
turnusername=28224511:1379330808
turnpassword=JZEOEt2V3Qb0y27GRntt2u2PAYA

then Asterisk 12 gets really slow and sometimes unresponsive. Calls result in 480 request timeout (possibly due to the freeze of Asterisk).

So this is also no solution.

Can not even test if it brings me some audio in my webRTC calls.

(putting the above lines back in comment resolves the issue of Asterisk freeze. This is all EXTREMELY BUGGY !)

Asterisk 13 here I come (with very high expectations).

Kind regards.

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
Â Â Â Â Â Â Â http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Â http://lists.digium.com/mailman/listinfo/asterisk-users

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