thr3ads.net - asterisk users - [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ? [Aug 2016]

If this information is useful, please help other people find it:
Share via:

Jonas Kellens

2016-Aug-09 20:48 UTC

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

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 at myserver admin]# yum install uuid-devel libuuid-devel
[root at myserver admin]# ./configure --libdir=/usr/lib64
[root at myserver admin]# make menuselect
[root at 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 :

[Aug  9 22:12:00]   == WebSocket connection from '178.119.146.190:36940'
for protocol 'sip' accepted using version '13'
[Aug  9 22:12:00]     -- Registered SIP '770000wrtc' at 
178.119.146.190:36940
[Aug  9 22:12:00]        > 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.


[Aug  9 22:15:50] <--- SIP read from WS:178.119.146.190:36940 --->
[Aug  9 22:15:50] INVITE sip:419 at 178.18.90.230 SIP/2.0
[Aug  9 22:15:50] Via: SIP/2.0/WSS 
df7jal23ls0d.invalid;branch=z9hG4bKk2KDePVLlTquEfJzIk7LCMdnOHHk4wn1;rport
[Aug  9 22:15:50] From: 
"77"<sip:770000wrtc at 178.18.90.230>;tag=sRCvFQq3gUMqkl6TKTcl
[Aug  9 22:15:50] To: <sip:419 at 178.18.90.230>
[Aug  9 22:15:50] Contact: 
"77"<sips:770000wrtc at
df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss>;+g.oma.sip-im;language="en,fr"
[Aug  9 22:15:50] Call-ID: 6aa0db27-a37b-69ee-8641-87c5bc444d32
[Aug  9 22:15:50] CSeq: 21553 INVITE
[Aug  9 22:15:50] Content-Type: application/sdp
[Aug  9 22:15:50] Content-Length: 1815
[Aug  9 22:15:50] Max-Forwards: 70
[Aug  9 22:15:50] Authorization: Digest 
username="770000wrtc",realm="178.18.90.230",nonce="1d8fa83d",uri="sip:419
at
178.18.90.230",response="cd2da8d1cbf0a2795b38b2048a3a3c49",algorithm=MD5
[Aug  9 22:15:50] User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
[Aug  9 22:15:50] Organization: Doubango Telecom
[Aug  9 22:15:50]
[Aug  9 22:15:50] v=0
[Aug  9 22:15:50] o=- 9108976588890881000 2 IN IP4 127.0.0.1
[Aug  9 22:15:50] s=Doubango Telecom - chrome
[Aug  9 22:15:50] t=0 0
[Aug  9 22:15:50] a=group:BUNDLE audio
[Aug  9 22:15:50] a=msid-semantic: WMS BJSlrOtzPj6wzI3QugifY58Oi18zpEbkNsps
[Aug  9 22:15:50] m=audio 41178 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 
105 13 126
[Aug  9 22:15:50] c=IN IP4 178.119.146.190
[Aug  9 22:15:50] a=rtcp:42197 IN IP4 178.119.146.190
[Aug  9 22:15:50] a=candidate:1668076467 1 udp 2122260223 192.168.1.122 
41178 typ host generation 0
[Aug  9 22:15:50] a=candidate:1668076467 2 udp 2122260222 192.168.1.122 
42197 typ host generation 0
[Aug  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
[Aug  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
[Aug  9 22:15:50] a=candidate:770649923 1 tcp 1518280447 192.168.1.122 0 
typ host tcptype active generation 0
[Aug  9 22:15:50] a=candidate:770649923 2 tcp 1518280446 192.168.1.122 0 
typ host tcptype active generation 0
[Aug  9 22:15:50] a=ice-ufrag:cd8nLIL1irEPdLZt
[Aug  9 22:15:50] a=ice-pwd:97awKXGiAt1TO5jlmb3GMXRy
[Aug  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
[Aug  9 22:15:50] a=setup:actpass
[Aug  9 22:15:50] a=mid:audio
[Aug  9 22:15:50] a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
[Aug  9 22:15:50] a=extmap:3 
http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
[Aug  9 22:15:50] a=sendrecv
[Aug  9 22:15:50] a=rtcp-mux
[Aug  9 22:15:50] a=rtpmap:111 opus/48000/2
[Aug  9 22:15:50] a=fmtp:111 minptime=10; useinbandfec=1
[Aug  9 22:15:50] a=rtpmap:103 ISAC/16000
[Aug  9 22:15:50] a=rtpmap:104 ISAC/32000
[Aug  9 22:15:50] a=rtpmap:9 G722/8000
[Aug  9 22:15:50] a=rtpmap:0 PCMU/8000
[Aug  9 22:15:50] a=rtpmap:8 PCMA/8000
[Aug  9 22:15:50] a=rtpmap:106 CN/32000
[Aug  9 22:15:50] a=rtpmap:105 CN/16000
[Aug  9 22:15:50] a=rtpmap:13 CN/8000
[Aug  9 22:15:50] a=rtpmap:126 telephone-event/8000
[Aug  9 22:15:50] a=maxptime:60
[Aug  9 22:15:50] a=ssrc:1885999682 cname:yLxCKvQLz0YJGRkR
[Aug  9 22:15:50] a=ssrc:1885999682 
msid:BJSlrOtzPj6wzI3QugifY58Oi18zpEbkNsps 
f0144e6c-86a1-4b08-bf58-4ced92361250
[Aug  9 22:15:50] a=ssrc:1885999682 
mslabel:BJSlrOtzPj6wzI3QugifY58Oi18zpEbkNsps
[Aug  9 22:15:50] a=ssrc:1885999682 
label:f0144e6c-86a1-4b08-bf58-4ced92361250
[Aug  9 22:15:50] <------------->
[Aug  9 22:15:50] --- (13 headers 40 lines) ---
[Aug  9 22:15:50] Using INVITE request as basis request - 
6aa0db27-a37b-69ee-8641-87c5bc444d32
[Aug  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'
[Aug  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'
[Aug  9 22:15:51] Found peer '770000wrtc' for '770000wrtc' from 
178.119.146.190:36940
[Aug  9 22:15:51]   == Using SIP RTP TOS bits 184
[Aug  9 22:15:51]   == Using SIP RTP CoS mark 5
[Aug  9 22:15:51] Found RTP audio format 111
[Aug  9 22:15:51] Found RTP audio format 103
[Aug  9 22:15:51] Found RTP audio format 104
[Aug  9 22:15:51] Found RTP audio format 9
[Aug  9 22:15:51] Found RTP audio format 0
[Aug  9 22:15:51] Found RTP audio format 8
[Aug  9 22:15:51] Found RTP audio format 106
[Aug  9 22:15:51] Found RTP audio format 105
[Aug  9 22:15:51] Found RTP audio format 13
[Aug  9 22:15:51] Found RTP audio format 126
[Aug  9 22:15:51] Found unknown media description format opus for ID 111
[Aug  9 22:15:51] Found unknown media description format ISAC for ID 103
[Aug  9 22:15:51] Found unknown media description format ISAC for ID 104
[Aug  9 22:15:51] Found audio description format G722 for ID 9
[Aug  9 22:15:51] Found audio description format PCMU for ID 0
[Aug  9 22:15:51] Found audio description format PCMA for ID 8
[Aug  9 22:15:51] Found unknown media description format CN for ID 106
[Aug  9 22:15:51] Found unknown media description format CN for ID 105
[Aug  9 22:15:51] Found audio description format CN for ID 13
[Aug  9 22:15:51] Found audio description format telephone-event for ID 126
[Aug  9 22:15:51] Capabilities: us - (alaw), peer - 
audio=(ulaw|alaw|g722)/video=(nothing)/text=(nothing), combined - (alaw)
[Aug  9 22:15:51] Non-codec capabilities (dtmf): us - 0x1 
(telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 
(telephone-event|)
[Aug  9 22:15:51] Peer audio RTP is at port 178.119.146.190:41178
[Aug  9 22:15:51] Looking for 419 in testwebrtc (domain 178.18.90.230)
[Aug  9 22:15:51] list_route: hop: 
<sips:770000wrtc at
df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss>
[Aug  9 22:15:51]
[Aug  9 22:15:51] <--- Transmitting (NAT) to 178.119.146.190:36940 --->
[Aug  9 22:15:51] SIP/2.0 100 Trying
[Aug  9 22:15:51] Via: SIP/2.0/WSS 
df7jal23ls0d.invalid;branch=z9hG4bKk2KDePVLlTquEfJzIk7LCMdnOHHk4wn1;received=178.119.146.190;rport=36940
[Aug  9 22:15:51] From: 
"77"<sip:770000wrtc at 178.18.90.230>;tag=sRCvFQq3gUMqkl6TKTcl
[Aug  9 22:15:51] To: <sip:419 at 178.18.90.230>
[Aug  9 22:15:51] Call-ID: 6aa0db27-a37b-69ee-8641-87c5bc444d32
[Aug  9 22:15:51] CSeq: 21553 INVITE
[Aug  9 22:15:51] Server: myPBX
[Aug  9 22:15:51] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, 
SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Aug  9 22:15:51] Supported: replaces, timer
[Aug  9 22:15:51] Contact: <sip:419 at 178.18.90.230:5060;transport=WS>
[Aug  9 22:15:51] Content-Length: 0
[Aug  9 22:15:51]
[Aug  9 22:15:51]
[Aug  9 22:15:51] <------------>
[Aug  9 22:15:51]     -- Executing [419 at testwebrtc:1] 
NoOp("SIP/770000wrtc-00000002", "") in new stack
[Aug  9 22:15:51]     -- Executing [419 at testwebrtc:4] 
Dial("SIP/770000wrtc-00000002", "SIP/testacc7700905") in new
stack
[Aug  9 22:15:51]   == Using SIP RTP TOS bits 184
[Aug  9 22:15:51]   == Using SIP RTP CoS mark 5
[Aug  9 22:15:51]     -- Called SIP/testacc7700905
[Aug  9 22:15:51]     -- SIP/testacc7700905-00000003 is ringing
[Aug  9 22:15:51]
[Aug  9 22:15:51] <--- Transmitting (NAT) to 178.119.146.190:36940 --->
[Aug  9 22:15:51] SIP/2.0 180 Ringing
[Aug  9 22:15:51] Via: SIP/2.0/WSS 
df7jal23ls0d.invalid;branch=z9hG4bKk2KDePVLlTquEfJzIk7LCMdnOHHk4wn1;received=178.119.146.190;rport=36940
[Aug  9 22:15:51] From: 
"77"<sip:770000wrtc at 178.18.90.230>;tag=sRCvFQq3gUMqkl6TKTcl
[Aug  9 22:15:51] To: <sip:419 at 178.18.90.230>;tag=as50efde9f
[Aug  9 22:15:51] Call-ID: 6aa0db27-a37b-69ee-8641-87c5bc444d32
[Aug  9 22:15:51] CSeq: 21553 INVITE
[Aug  9 22:15:51] Server: myPBX
[Aug  9 22:15:51] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, 
SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Aug  9 22:15:51] Supported: replaces, timer
[Aug  9 22:15:51] Contact: <sip:419 at 178.18.90.230:5060;transport=WS>
[Aug  9 22:15:51] Content-Length: 0
[Aug  9 22:15:51]
[Aug  9 22:15:51]
[Aug  9 22:15:51] <------------>
[Aug  9 22:15:51]     -- SIP/testacc7700905-00000003 is ringing
[Aug  9 22:15:52]        > 0x7fc5dc014060 -- Probation passed - setting 
RTP source address to 178.119.159.58:44704
[Aug  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'
[Aug  9 22:15:52]     -- SIP/testacc7700905-00000003 answered 
SIP/770000wrtc-00000002
[Aug  9 22:15:52] Audio is at 11536
[Aug  9 22:15:52] Adding codec 100004 (alaw) to SDP
[Aug  9 22:15:52] Adding non-codec 0x1 (telephone-event) to SDP
[Aug  9 22:15:52]
[Aug  9 22:15:52] <--- Reliably Transmitting (NAT) to 
178.119.146.190:36940 --->
[Aug  9 22:15:52] SIP/2.0 200 OK
[Aug  9 22:15:52] Via: SIP/2.0/WSS 
df7jal23ls0d.invalid;branch=z9hG4bKk2KDePVLlTquEfJzIk7LCMdnOHHk4wn1;received=178.119.146.190;rport=36940
[Aug  9 22:15:52] From: 
"77"<sip:770000wrtc at 178.18.90.230>;tag=sRCvFQq3gUMqkl6TKTcl
[Aug  9 22:15:52] To: <sip:419 at 178.18.90.230>;tag=as50efde9f
[Aug  9 22:15:52] Call-ID: 6aa0db27-a37b-69ee-8641-87c5bc444d32
[Aug  9 22:15:52] CSeq: 21553 INVITE
[Aug  9 22:15:52] Server: myPBX
[Aug  9 22:15:52] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, 
SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Aug  9 22:15:52] Supported: replaces, timer
[Aug  9 22:15:52] Contact: <sip:419 at 178.18.90.230:5060;transport=WS>
[Aug  9 22:15:52] Content-Type: application/sdp
[Aug  9 22:15:52] Content-Length: 387
[Aug  9 22:15:52]
[Aug  9 22:15:52] v=0
[Aug  9 22:15:52] o=myPBX 1420513531 1420513531 IN IP4 178.18.90.230
[Aug  9 22:15:52] s=myPBX
[Aug  9 22:15:52] c=IN IP4 178.18.90.230
[Aug  9 22:15:52] t=0 0
[Aug  9 22:15:52] m=audio 11536 RTP/SAVPF 8 126
[Aug  9 22:15:52] a=rtpmap:8 PCMA/8000
[Aug  9 22:15:52] a=rtpmap:126 telephone-event/8000
[Aug  9 22:15:52] a=fmtp:126 0-16
[Aug  9 22:15:52] a=ptime:20
[Aug  9 22:15:52] a=connection:new
[Aug  9 22:15:52] a=setup:active
[Aug  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
[Aug  9 22:15:52] a=sendrecv
[Aug  9 22:15:52]
[Aug  9 22:15:52] <------------>
[Aug  9 22:15:52]
[Aug  9 22:15:52] <--- SIP read from WS:178.119.146.190:36940 --->
[Aug  9 22:15:52] ACK sip:419 at 178.18.90.230:5060;transport=WS SIP/2.0
[Aug  9 22:15:52] Via: SIP/2.0/WSS 
df7jal23ls0d.invalid;branch=z9hG4bKe2fFxswvLSg8fovfxEpP;rport
[Aug  9 22:15:52] From: 
"77"<sip:770000wrtc at 178.18.90.230>;tag=sRCvFQq3gUMqkl6TKTcl
[Aug  9 22:15:52] To: <sip:419 at 178.18.90.230>;tag=as50efde9f
[Aug  9 22:15:52] Contact: 
"77"<sips:770000wrtc at
df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss>;+g.oma.sip-im;language="en,fr"
[Aug  9 22:15:52] Call-ID: 6aa0db27-a37b-69ee-8641-87c5bc444d32
[Aug  9 22:15:52] CSeq: 21553 ACK
[Aug  9 22:15:52] Content-Length: 0
[Aug  9 22:15:52] Max-Forwards: 70
[Aug  9 22:15:52] Authorization: Digest 
username="770000wrtc",realm="178.18.90.230",nonce="1d8fa83d",uri="sip:419
at
178.18.90.230:5060;transport=WS",response="fb65d05b7872c6650836d83535122ef1",algorithm=MD5
[Aug  9 22:15:52] User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
[Aug  9 22:15:52] Organization: Doubango Telecom
[Aug  9 22:15:52]
[Aug  9 22:15:52] <------------->
[Aug  9 22:15:52] --- (12 headers 0 lines) ---
[Aug  9 22:15:52]        > 0x7fc5dc014060 -- Probation passed - setting 
RTP source address to 178.119.159.58:44704
[Aug  9 22:15:52]        > 0x7fc5dc014060 -- Probation passed - setting 
RTP source address to 178.119.159.58:44704



[Aug  9 22:17:08] Sent RTP packet to      178.119.146.190:59051 (type 
08, seq 028865, ts 2789673216, len 000160)
[Aug  9 22:17:09] Sent RTP packet to      178.119.146.190:59051 (type 
08, seq 028866, ts 2789673376, len 000160)
[Aug  9 22:17:09] Sent RTP packet to      178.119.146.190:59051 (type 
08, seq 028867, ts 2789673536, len 000160)
[Aug  9 22:17:09] Sent RTP packet to      178.119.146.190:59051 (type 
08, seq 028868, ts 2789673696, len 000160)
[Aug  9 22:17:09] Sent RTP packet to      178.119.146.190:59051 (type 
08, seq 028869, ts 2789673856, len 000160)
[Aug  9 22:17:09] Sent RTP packet to      178.119.146.190:59051 (type 
08, seq 028870, ts 2789674016, len 000160)
[Aug  9 22:17:09] Sent RTP packet to      178.119.146.190:59051 (type 
08, seq 028871, ts 2789674176, len 000160)
[Aug  9 22:17:09] Sent RTP packet to      178.119.146.190:59051 (type 
08, seq 028872, ts 2789674336, len 000160)
[Aug  9 22:17:09] Sent RTP packet to      178.119.146.190:59051 (type 
08, seq 028873, ts 2789674496, len 000160)
[Aug  9 22:17:09] Sent RTP packet to      178.119.146.190:59051 (type 
08, seq 028874, ts 2789674656, len 000160)
[Aug  9 22:17:09] Sent RTP packet to      178.119.146.190:59051 (type 
08, seq 028875, ts 2789674816, len 000160)
[Aug  9 22:17:09] Sent RTP packet to      178.119.146.190:59051 (type 
08, seq 028876, ts 2789674976, len 000160)
[Aug  9 22:17:09] Sent RTP packet to      178.119.146.190:59051 (type 
08, seq 028877, ts 2789675136, len 000160)
[Aug  9 22:17:09] Sent RTP packet to      178.119.146.190:59051 (type 
08, seq 028878, ts 2789675296, len 000160)
[Aug  9 22:17:09] Sent RTP packet to      178.119.146.190:59051 (type 
08, seq 028879, ts 2789675456, len 000160)
[Aug  9 22:17:09] Sent RTP packet to      178.119.146.190:59051 (type 
08, seq 028880, ts 2789675616, len 000160)
[Aug  9 22:17:09] Sent RTP packet to      178.119.146.190:59051 (type 
08, seq 028881, ts 2789675776, len 000160)
[Aug  9 22:17:09] Sent RTP packet to      178.119.146.190:59051 (type 
08, seq 028882, ts 2789675936, len 000160)
[Aug  9 22:17:09] Sent RTP packet to      178.119.146.190:59051 (type 
08, seq 028883, ts 2789676096, len 000160)
[Aug  9 22:17:09] Sent RTP packet to      178.119.146.190:59051 (type 
08, seq 028884, ts 2789676256, len 000160)
[Aug  9 22:17:09] Sent RTP packet to      178.119.146.190:59051 (type 
08, seq 028885, ts 2789676416, len 000160)
[Aug  9 22:17:09] Sent RTP packet to      178.119.146.190:59051 (type 
08, seq 028886, ts 2789676576, len 000160)
[Aug  9 22:17:09] Sent RTP packet to      178.119.146.190:59051 (type 
08, seq 028887, ts 2789676736, len 000160)
[Aug  9 22:17:09] Sent RTP packet to      178.119.146.190:59051 (type 
08, seq 028888, ts 2789676896, len 000160)
[Aug  9 22:17:09] Sent RTP packet to      178.119.146.190:59051 (type 
08, seq 028889, ts 2789677056, len 000160)
[Aug  9 22:17:09] Sent RTP packet to      178.119.146.190:59051 (type 
08, seq 028890, ts 2789677216, len 000160)
[Aug  9 22:17:09] Sent RTP packet to      178.119.146.190:59051 (type 
08, seq 028891, ts 2789677376, len 000160)
[Aug  9 22:17:09] Sent RTP packet to      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.

-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20160809/056732b6/attachment.html>

Ludovic Gasc

2016-Aug-10 06:52 UTC

head link

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

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

Have a nice day.

Ludovic Gasc (GMLudo)
http://www.gmludo.eu/
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20160810/5d17171f/attachment.html>

Jonas Kellens

2016-Aug-10 10:02 UTC

head link

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

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.

-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20160810/eb12407f/attachment.html>

Seemingly Similar Threads

Search for more maybe matching threads

asterisk users - Aug 2016 - Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

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

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

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

Seemingly Similar Threads