Cloud CM-IPMP Guide de dépannage Page 154

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address-of-record = sip:[email protected]
Updating bindings
Updating binding: sip:[email protected] -> sip:[email protected]
Contact: <sip:[email protected]>
setRegistrationTimer(sip:[email protected], sip:[email protected], 900, 2400921797@192.
168.0 9, 0)
set new timer for registration: sip:[email protected] -> sip:[email protected], expires
in 900s
Adding 1 headers
Sending Response:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.9:5060;branch=z9hG4bK4101313487
From: <sip:[email protected]>;tag=1555020692
To: <sip:[email protected]>;tag=1555020692
CSeq: 0 REGISTER
Max-Forwards: 10
Contact: <sip:[email protected]>;expires=900;q=0.0
Date: Sun, 18 Apr 2004 10:55:23 GMT
Content-Length: 0
Note that the first REGISTER request processed by the SLEE after it starts up may take slightly longer than normal. This is
due to one-time initialisation of some SIP stack and SLEE classes. Subsequent requests will be much quicker.
22.6.3 Using the Proxy Service
The SIP Proxy Service can be used to setup a call between two Linphone user agents on the same network. The Proxy Service
does not support advanced features like authentication or request forking, and can only be used within a single domain.
It is necessary to run the Linphone user agents on separate hosts. This is so that the RTP ports used by Linphone for audio data
do not conflict. Assume the two hosts in our example are called
siptest1
and
siptest2
.
Rhino SLEE may run on one of these hosts (assume siptest1) as long as the SIP RA UDP port (default 5060) does not conflict
with the Linphone SIP port on that host.
On each host, setup the Linphone user agent to use siptest1 as the SIP server. Configure the user agent on siptest1 to use the
address-of-record
and the siptest2 user agent to use the address-of-record
.
Start both user agents. Both should register automatically with the Registrar service (the Registrar service is installed with the
Proxy service as a “Child SBB”).
Once both agents have registered, it is then possible to make a call to a user’s public SIP address via the Proxy service.
The Proxy service will retrieve the callee’s contact address from the Location Service and route the call (a SIP INVITE request)
to the destination user agent.
On siptest1 (Joe’s host), enter the SIP address
, and press
Call or Answer
. This will send a SIP
INVITE request to Fred, via the Proxy Service.
The status bars on the user agents should show the call in progress. A ringing tone will be heard if sound is enabled. On siptest2,
hit
Call or Answer
to accept the call. This will complete the INVITE-200 OK-ACK SIP handshake and setup the call. Both
user agents should now show
Connected
in the status bar.
If the local systems have microphone inputs enabled then it should be possible to speak to the other party. The audio data is
transferred directly between the user agents over a separate RTP connection. This connection is not managed by the SIP proxy
service.
Either user can then hangup the call by hitting
Release or Refuse
. This will send a SIP BYE request to the other user agent.
Hitting
Release or Refuse
on the caller while the INVITE is in progress will send a CANCEL request. If the callee hits
Release or Refuse
, this will cause a
603 Decline
response to be returned to the caller.
Open Cloud Rhino 1.4.3 Administration Manual v1.1 145
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