[Freeswitch-users] Sending SAVPF INVITE to Opensips

[James Mortensen]

I wanted to follow up on Kristian's question about OpenSIPS and WebSocket
support.  According to what I've read, OpenSIPS only supports WS/WSS Via
header parsing but doesn't actually have a WebSocket module to accept
WebSocket connections.  According to the OpenSIPS documentation as well as
the release notes, one must front-end OpenSIPS with OverSIP.

http://www.opensips.org/Documentation/Tutorials-WebSocket

Hope this helps!


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=" <http://bandwidth.com>
>>
>>
>
> --
> Sent from mobile device
>
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