[Sipp-users] Creating a dialup Audio/Video loopback: Registering SIPp with Webex Control Hub, UAS and -rtp_echo

[Economides, T. <tec...@ve...>]

My goal here is to use SIPp in server mode, have it register to Webex Control Hub and wait for calls to be routed to it (via an internal extension). When calls come to SIPp it should echo back all RTP streams that it receives.  We use such a loopback scenario (using hardware videoconf systems, at the moment, to receive and loop back the calls) to test conference room videoconf systems, by letting them dial up the loopback and check that audio and video from the room is looped back.

Anyhow, I'm looking for anyone who's successfully gotten SIPp to talk to Webex Control Hub, using TLS encryption, that can give me some guidance.

Right now I am puzzled at how to set up both Control Hub and SIPp so that they actually talk to one another.  Here's what I've tried:


  *   In Control Hub (CH), I created a 'Generic IPPhone Customer Managed" device, and that device is contained within a Workspace with Webex Calling enabled. CH also allows for a 'Generic IPgateway Customer Managed" but I haven't tried using that configuration yet.)
  *   I've assigned an extension to the Workspace (x1277)
  *   CH provided a username and password, along with an outbound proxy server name and a 'LineID':
     *   Outbound Proxy is: jfk06.hosted-us.bcld.webex.com. CH docs state that this is DNS resolved with SRV records, and SIPp doesn't support SRV lookup from what I can tell. So I manually resolved this address to 23.89.40.55 and port 8934
     *   Line ID is of the form: <local-name-of-random-letters-and-numbers>@83549438.cisco-bcld.com
  *   UAS: Because I want SIPp to listen for calls and respond to them, it seems that SIPp should be in UAS mode. Right?
  *   My SIPp command line looks like: sudo ./sipp 23.89.40.55:8934 -i <local, NATted IP addy> -p 5060 -sn UAS -t 1n -max_socket 1023 -au <SIP username> -ap <SIP passwd> -s <LineID user part, left of @> -rtp_echo -trace_err -trace_msg -trace_error_codes -trace_calldebug -trace_logs -trace_screen
  *   When I run the command, I can see the Scenario Screen, and it shows that the REGISTER -----> messages quantity is 30 (I'll post a screen shot but I'm guessing it won't come through the listserve)
[cid:image001.png@01DB7EC7.4DE05220]

When I do this, there's no indication that SIPp is talking to CH.  I use netstat to watch for traffic on local port 5060 when I run SIPp, but no activity is detected.  Further, when I try dialing the 1277 extension from another device registered to CH, I get an error recording saying "we're sorry, your call cannot be completed."

There is a firewall between the local VM where I'm running SIPp and the Internet, but there's no blocking of packets showing up there at all.

I have a feeling that I'm trying to do something that SIPp is capable of (registering to another SIP server and waiting for calls to be forwarded to it) but I need some help.

Cheers!

Theo Economides, AV / Multimedia Supervisor

VedderPrice
T +1 312 609 4597
222 North LaSalle Street, Chicago, Illinois 60601
web<http://www.vedderprice.com/> | email<mailto:tec...@ve...> | offices<http://www.vedderprice.com/locations/>

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