[Freeswitch-users] Sending SAVPF INVITE to Opensips

[James Mortensen]

Hi Kristian,

You're right, what I want is a WebRTC SDP, which FreeSWITCH is clearly
capable of generating as it does this with the default configuration by
default if port 5066 is enabled for the ws-binding parameter.

To answer your question, OverSIP sits in front of Opensips between the
Chrome client and Opensips.  Chrome can only register with a SIP endpoint
using websockets, so OverSIP listens on a TCP/WS port and passes the SIP
messages to Opensips.

BANDWIDTH.com --udp-->  FreeSWITCH -udp-->  Opensips -udp--> OverSIP
--ws--> Chrome WebRTC client.

It looks like what I was missing was the <action application="export"
data="media_webrtc=true"/> and using application export instead of set.  I
was able to get an inbound call routed to my Chrome client, answer it, and
experience two way audio.


Thanks again! :)


James



On Sat, Sep 28, 2013 at 12:46 PM, Kristian Kielhofner <kris at kriskinc.com>wrote:

> James,
>
>   There's quite a bit of detail omitted here but a few points:
>
> - avpf=yes on your gateway definition isn't doing anything.
> - Have you tried exporting media_webrtc=true before bridging back to
> Opensips?
> - Try exporting the actual variables, not including variable_ :
>
> <extension name="bandwidth.com inbound bridge">
>     <condition field="destination_number" expression="^\+1(5035551212)$">
>        <action application="answer" /> <!-- This probably shouldn't be
> here either -->
>        <action application="export" data="sip_auth_username=11234"/>
>        <action application="export" data="sip_auth_password=password"/>
>        <action application="export" data="media_webrtc=true"/>
>        <action application="bridge"
> data="sofia/external/ws-Opensips/11234 at 54.X.X.75"/>
>     </condition>
>   </extension>
>
> - While you do want an "SAVPF INVITE" you really want a WebRTC SDP,
> which includes much much more than just SAVPF.
> - You'll probably want to do all other sorts of codec manipulation,
> fixups, etc when bridging between typical SIP endpoints and WebRTC
> endpoints.  Look into late negotiation and every codec
> variable/setting you can find.
> - You may want to re-consider the interaction and authentication
> between OpenSIPS and FreeSWITCH.
>
>   Also, what are you using OverSIP for in this scenario?
>
>
> On Fri, Sep 27, 2013 at 8:26 PM, James Mortensen
> <james.mortensen at synclio.com> wrote:
> > Hello,
> >
> > I have a bandwidth.com number pointed to opensips, and a WebRTC peer
> > registered with Opensips.  I'm trying to dial the 10 digit number from a
> > cell phone and connect the call through FreeSWITCH to the Chrome WebRTC
> > client.
> >
> >
> > I defined opensips as a gateway, in the external profile:
> >
> > <include>
> >    <gateway name="ws-Opensips">
> >      <!-- <param name="from-user" value="fromuser"/> -->
> >      <param name="from-domain" value="54.X.X.75"/>
> >      <param name="proxy" value="54.X.X.75"/>
> >      <param name="expire-seconds" value="600"/>
> >      <param name="register" value="false"/>
> >      <param name="retry_seconds" value="30"/>
> >      <param name="extension" value="18257773456"/>
> >      <param name="context" value="public"/>
> >      <param name="avpf" value="yes"/>
> >      <param name="username" value="11234"/>
> >      <param name="password" value="password"/>
> >    </gateway>
> > </include>
> >
> >
> > In the public dialplan context, I added in a condition to catch the
> INVITE
> > coming in from opensips and pass it to a context I've called
> > "default-inbound". See the second condition:
> >
> >  <extension name="from_opensips">
> >     <condition field="network_addr" expression="^54\.X\.X\.75$"
> > break="never"> <!--CUSTOMIZE-->
> >       <action application="transfer" data="${destination_number} XML
> > default"/>
> >     </condition>
> >     <condition field="network_addr" expression="^54\.X\.X\.111$">
> > <!--CUSTOMIZE Use a third context here -->
> >       <action application="transfer" data="${destination_number} XML
> > default-inbound"/>
> >     </condition>
> >   </extension>
> >
> >
> > Then, in the default-inbound context, I match the dialed number, answer
> the
> > call leg from the PSTN, and then try to transfer back through opensips to
> > oversip and to Chrome.  The problem is that I either end up sending back
> AVP
> > INVITES, or Opensips refuses to authenticate the user.
> >
> > <extension name="bandwidth.com inbound bridge">
> >     <condition field="destination_number" expression="^\+1(5035551212)$">
> >        <action application="answer" />
> >        <action application="set"
> data="variable_sip_auth_username=11234"/>
> >        <action application="set"
> > data="variable_sip_auth_password=password"/>
> >        <action application="bridge"
> > data="sofia/external/ws-Opensips/11234 at 54.X.X.75"/>
> >
> >     </condition>
> >   </extension>
> >
> >
> > As you can see, I've hard-coded my peer, 11234, in the configuration.
> This
> > is a registered user on Opensips.
> >
> > How can I get FreeSWITCH to send the SAVPF INVITE through to Opensips for
> > the WebRTC portion of the call leg?  I apologize if this is covered
> > somewhere, but I've been wracking my brain on this for days and am not
> > getting anywhere.
> >
> > The Opensips configuration I have works with existing Asterisk 11
> servers,
> > and I'm hoping I can just simply plug in FreeSWITCH servers seamlessly
> into
> > the mix.
> >
> > Thank you!
> >
> > James
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
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>
>
> --
> Kristian Kielhofner
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
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