[Freeswitch-users] Sending SAVPF INVITE to Opensips

[Michael Jerris]

Were you aware that FreeSWITCH also supports websockets natively?

On Sep 29, 2013, at 1:01 PM, James Mortensen <james.mortensen at synclio.com> wrote:

> Hi Kristian, 
> 
> No, I wasn't aware OpenSIPS handled WebSockets. Looks like it's even been supported since v1.9, which is the one we're using.
> 
> This gives us more to think about.  Thank you!
> 
> 
> James
> 
> 
> 
> On Sat, Sep 28, 2013 at 4:22 PM, Kristian Kielhofner <kris at kriskinc.com> wrote:
> I'm glad it worked for you.
> 
> Are you aware that Opensips supports SIP over websockets and secure websockets natively?
> 
> 
> On Saturday, September 28, 2013, James Mortensen wrote:
> 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="
> 
> 
> -- 
> Sent from mobile device
> 
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