[asterisk-users] Asterisk removes ice lines in sdp when calling between webrtc clients

[Matthew Jordan]

On Mon, Sep 8, 2014 at 9:48 AM, Olli Heiskanen <
ohjelmistoarkkitehti at gmail.com> 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:660 at PU.BL.IC.IP: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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