[Freeswitch-users] Sending SAVPF INVITE to Opensips

[James Mortensen]

Hi Stanislav,

Opensips doesn't register our media servers, it just passed traffic through
it and is all based on IP address.  What I'm trying to do is get FreeSWITCH
to take an AVP INVITE from Bandwidth.com, answer that call leg, then send
an SAVPF INVITE to Opensips then to OverSIP and finally to a Chrome client.
 Instead, it's sending an AVP INVITE.

Hence, the INVITE reaches my Chrome client endpoint, but Chrome responds,
rightly so, with a 488 Not Acceptable Here because Chrome only supports
SAVPF profiles.

In Asterisk, in the gateway configuration, I would put avpf=yes to tell
Asterisk to send the INVITE with SAVPF, but on FreeSWITCH, since WebRTC is
so new, and since I'm only about a week into my experiences with
FreeSWITCH, the documentation for this is tough to find.

Hope this helps clarify, and thanks for responding.


James


On Sat, Sep 28, 2013 at 3:29 AM, Stanislav Sinyagin <ssinyagin at yahoo.com>wrote:

> in your ws-Opensips gateway definition, you have "register" set to "false"
> If you change it to "true", FreeSWITCH will authenticate on that gateway.
>
> is that what you try to achieve?
>
>
>
>   ------------------------------
>  *From:* James Mortensen <james.mortensen at synclio.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Saturday, September 28, 2013 2:26 AM
> *Subject:* [Freeswitch-users] Sending SAVPF INVITE to Opensips
>
> 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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> http://www.freeswitchsolutions.com
>
> 
> 
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>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
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