asterisk users - Aug 2014 - Letting rtp profiles be handled by rtpengine instead of Asterisk

Hello,

I'm trying to get calls working between websocket clients and sip clients.
For clients I have sip.js based clients on chrome, Zoipers and a
Grandstream phone. Challenge here is I'd like to have Kamailio and
rtpengine to handle the bridging between different rtp profiles but
Asterisk changes them in the sdp bodies along the way. I'm using Asterisk
11.11.0.

Is there a way to configure Asterisk to ignore the rtp profile but allow
calls to pass with either of those profiles (even though clients might
answer with 488 which would be caught and handled by Kamailio and
rtpengine)? In my setup I have Asterisk Kamailio realtime integration, and
the second goal is to be able to add peers to the db table with similar
data, as in no different values based on what kind of client wants to
register. I'd like to allow the user to register using which ever client
they choose (in this case one of the 3 I mentioned).

Previously I had problems like 'rejecting secure audio stream without
encryption details', no audio or BYE messages sent immediately after call
has begun etc, but according to sip.js documentation (
http://sipjs.com/guides/server-configuration/asterisk/) the settings avpf
and force_avp affect the way Asterisk handles the rtp profiles and now my
calls do work ok but I'd need to move the rtp profile handling to rtpengine.

Here's my sip.conf:

bindport = 5070 ;Kamailio is at port 5060, and it's always used as outbound
proxy
bindaddr = PU.BL.IC.IP
tcpenable = yes
limitonpeers = yes
rtcachefriends = yes
rtupdate=yes
tos_sip=cs3
tos_audio=ef
realm = testers.com
 autodomain=no
domain=testers.com

allowexternaldomains=no
allowguest=no
;avpf=yes ;
encryption=yes
transport=ws,wss,udp
icesupport=yes
srvlookup=yes
nat=force_rport,comedia
videosupport=yes
directmedia=no


And here's the way I've defined my websocket peer to my sippeers table:

            id: 4
          name: 660
        ipaddr: PU.BL.IC.IP
          port: 5060
    regseconds: 1407744248
   defaultuser: 660
   fullcontact: sip:660 at PU.BL.IC.IP:5060
     regserver:
     useragent:
        lastms: 0
          host: dynamic
          type: friend
       context: default
          deny: 0.0.0.0/0.0.0.0
        permit: PU.BL.IC.IP
        secret: NULL
     md5secret: NULL
          avpf: yes
     force_avp: yes
    icesupport: yes
   directmedia: yes
    encryption: yes
           nat: force_rport,comedia
     callgroup: NULL
   pickupgroup: NULL
      language: NULL
      disallow: NULL
         allow: NULL
        setvar: NULL
      callerid: NULL
      amaflags: NULL
  videosupport: no
maxcallbitrate: NULL
       mailbox: NULL
      regexten: NULL
    fromdomain: testers.com
      fromuser: NULL
        qualify: NULL
     defaultip: NULL
 outboundproxy: PU.BL.IC.IP
 contactpermit: NULL
   contactdeny: NULL
      fullname: NULL
    cid_number: NULL
   callingpres: NULL
  mohinterpret: NULL
    mohsuggest: NULL
  hasvoicemail: NULL
  subscribemwi: NULL
       vmexten: NULL
  rtpkeepalive: NULL
directrtpsetup: yes
    dtlsenable: yes
    dtlsverify: no
dtlsprivatekey: /etc/asterisk/keys/asterisk.pem
     dtlssetup: actpass
  dtlscertfile: /etc/asterisk/keys/asterisk.pem
    dtlscafile: /etc/asterisk/keys/ca.crt
     sippasswd: md5ofmypwd
          rpid: NULL
        domain: testers.com
    sippasswd2: NULL



This is how all other clients are currently defined:

            id: 7
          name: 771
        ipaddr: PU.BL.IC.IP
          port: 5060
    regseconds: 1407748788
   defaultuser: 771
   fullcontact: sip:771 at PU.BL.IC.IP:5060
     regserver:
     useragent:
        lastms: 0
          host: dynamic
          type: friend
       context: default
          deny: 0.0.0.0/0.0.0.0
        permit: PU.BL.IC.IP
        secret: NULL
     md5secret: NULL
          avpf: no
     force_avp: NULL
    icesupport: NULL
   directmedia: yes
    encryption: NULL
           nat: force_rport,comedia
     callgroup: NULL
   pickupgroup: NULL
      language: NULL
      disallow: NULL
         allow: NULL
        setvar: NULL
      callerid: NULL
      amaflags: NULL
  videosupport: NULL
maxcallbitrate: NULL
       mailbox: NULL
      regexten: NULL
    fromdomain: testers.com
      fromuser: NULL
       qualify: NULL
     defaultip: NULL
 outboundproxy: PU.BL.IC.IP
 contactpermit: NULL
   contactdeny: NULL
      fullname: NULL
    cid_number: NULL
   callingpres: NULL
  mohinterpret: NULL
    mohsuggest: NULL
  hasvoicemail: NULL
  subscribemwi: NULL
       vmexten: NULL
  rtpkeepalive: NULL
directrtpsetup: NULL
    dtlsenable: NULL
    dtlsverify: NULL
dtlsprivatekey: NULL
     dtlssetup: NULL
  dtlscertfile: NULL
    dtlscafile: NULL
     sippasswd: 27e13af7c596313350986c58c9d24946
          rpid: NULL
        domain: testers.com
    sippasswd2: NULL


cheers,
Olli
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On Mon, Aug 11, 2014 at 4:45 AM, Olli Heiskanen
<ohjelmistoarkkitehti at gmail.com> wrote:
>
> Hello,
>
> I'm trying to get calls working between websocket clients and sip
clients.
> For clients I have sip.js based clients on chrome, Zoipers and a
Grandstream
> phone. Challenge here is I'd like to have Kamailio and rtpengine to
handle
> the bridging between different rtp profiles but Asterisk changes them in
the
> sdp bodies along the way. I'm using Asterisk 11.11.0.
>
> Is there a way to configure Asterisk to ignore the rtp profile but allow
> calls to pass with either of those profiles (even though clients might
> answer with 488 which would be caught and handled by Kamailio and
> rtpengine)? In my setup I have Asterisk Kamailio realtime integration, and
> the second goal is to be able to add peers to the db table with similar
> data, as in no different values based on what kind of client wants to
> register. I'd like to allow the user to register using which ever
client
> they choose (in this case one of the 3 I mentioned).
>
> Previously I had problems like 'rejecting secure audio stream without
> encryption details', no audio or BYE messages sent immediately after
call
> has begun etc, but according to sip.js documentation
> (http://sipjs.com/guides/server-configuration/asterisk/) the settings avpf
> and force_avp affect the way Asterisk handles the rtp profiles and now my
> calls do work ok but I'd need to move the rtp profile handling to
rtpengine.
>
We are successfully using kamailio / rtpengine with websockets and
asterisk 1.8. First question is why are you duplicating registrations
within asterisk?  Secondly, why are you using websockets in asterisk?

Without knowing more about your use case, I'll tell you how we did it.
Like I said, kamailio is responsible for our SIP/ws subscribers and
registrations.  Once within kamailio we simply dispatch traffic to
asterisk via SIP/udp.  RTP is handled by rtpengine (using rtproxy-ng)
and that is basically it.

No special configuration is needed for asterisk (in fact 1.8 has no
support for RTP/SAVPF) so we rewrite SDP on 488.  Then setup a
kamailio peer and away you go.

-- 
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belanger at polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger