asterisk users - Sep 2014 - Asterisk removes ice lines in sdp when calling between webrtc clients

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.

cheers,
Olli
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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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