[Freeswitch-users] Sending SAVPF INVITE to Opensips

[Kristian Kielhofner]

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
>
>
>
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-- 
Kristian Kielhofner