On Mon, Sep 8, 2014 at 9:48 AM, Olli Heiskanen < [email protected]> wrote:
> Hello, > > I have a problem with a call between 2 webrtc clients. Asterisk removes > the ice-related lines from the sdp when it sends the INVITE out, and the > called webrtc client rejects the INVITE due to the missing ice lines. Both > webrtc clients are defined exactly the same way, same values in all fields > except the number of the peer. > > There's probably something I've changed that causes this behavior. Can > anyone tell me what's wrong in my configuration? > > res_rtp_asterisk is included in the compilation and uuid-devel is > installed, Asterisk version is 11.11.0. Ice is enabled in rtp.conf as well > as in both clients in the realtime sip peer table. > > Here's my realtime peer data: > *CLI> realtime load sippeers name 660 > Column Name Column Value > -------------------- -------------------- > id 4 > type friend > name 660 > host dynamic > secret > encryption yes > avpf yes > icesupport yes <---- ICE is enabled > ipaddr PU.BL.IC.IP > port 5060 > regseconds 1410185500 > defaultuser 660 > fullcontact sip:[email protected]:5060 > lastms 0 > useragent > context default > directmedia no > deny 0.0.0.0/0.0.0.0 > permit PU.BL.IC.IP > nat force_rport,comedia > language > disallow > allow > force_avp yes > callerid > amaflags > mailbox > regexten > regserver > fromdomain testers.com > videosupport no > contactpermit > contactdeny > fullname 660 win8 > hasvoicemail > subscribemwi > dtlsenable yes > dtlsverify no > dtlscertfile /etc/asterisk/keys/asterisk.pem > dtlsprivatekey /etc/asterisk/keys/asterisk.pem > dtlssetup actpass > sippasswd md5pwd > rpid > domain testers.com > sippasswd2 > > and my sip.conf: > > [general] > bindport = 5070 > bindaddr = PU.BL.IC.IP > udpbindaddr = PU.BL.IC.IP > tcpenable = yes > limitonpeers = yes > rtcachefriends = no > tos_sip=cs3 > tos_audio=ef > realm = testers.com > autodomain=yes > domain=PU.BL.IC.IP > domain=testers.com > transport=ws,wss,udp > outboundproxy=PU.BL.IC.IP:5060 > > > I'd appreciate Your advice. > > > What does a DEBUG log show with 'sip set debug on' when the outbound call is made? -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org
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