[Serusers] Re: Serusers Digest, Vol 13, Issue 2 - sr-users

2 May 2004

      Arnd,
You may use SER MediaProxy if you wish real-time statistics and 
accounting of network traffic.
See some captured example:
http://www.ag-projects.com/MediaSessions.html
http://www.ag-projects.com/CDRTool.html
Adrian
On May 2, 2004, at 12:00 PM, serusers-request@lists.iptel.org wrote:
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Today's Topics:

Any way to list rtpproy bindings? (Arnd Vehling)
Voice loss after establishing call (Arnd Vehling)

Message: 1
Date: Sat, 01 May 2004 15:22:08 +0200
From: Arnd Vehling av@nethead.de
Subject: [Serusers] Any way to list rtpproy bindings?
To: SER Mailing List serusers@lists.iptel.org
Message-ID: 4093A480.4070508@nethead.de
Content-Type: text/plain; charset=us-ascii; format=flowed
Hi,
is there any way to list the current rtp-proxy bindings?
Searching through the source code ive found that rtpproxy
understands a "L" or "l" command to list the current bindings.
I tried to "echo "L" > /var/run/rtpproxy.sock" which results in
an error and i tried to communicate with ser via a perl-script
(see below) to no avail. Could it be that rtpproxy can only
communicate with one process a time? Is there any other way
to dump the list of udp bindings?
best regards,
Arnd
#!/usr/bin/perl
use IO::Socket;
$client = IO::Socket::UNIX->new(PeerAddr  => "/var/run/rtpproxy.sock",
                                 Type      => SOCK_DGRAM,
                                 Timeout   => 10 )
     or die $@;
print $client "L\n";
$answer = <$client>;
print "proxy said: $answer\n";
close( $client );

Message: 2
Date: Sat, 01 May 2004 16:05:03 +0200
From: Arnd Vehling av@nethead.de
Subject: [Serusers] Voice loss after establishing call
To: SER Mailing List serusers@lists.iptel.org
Message-ID: 4093AE8F.1090804@nethead.de
Content-Type: text/plain; charset=us-ascii; format=flowed
Hello,
we currently experience "incoming voice loss" after an outgoing
call has been established. i.e. after the remote party picks up the
call you dont hear anything for 1-2 seconds which results in both
sides of the connection saying repeatedly "Hello" until both sides
can hear each other.
This phenomenon occurs with nated and non natted clients so it
doesnt seem to be related to "rtpproxy" or other NAT problems.
It occurs between  SIP<>SIP and SIP<>PSTN Calls (via a cisco gw)
independantly of the UA type used.
Has anyone an idea where to look for the bug/where to start debugging?
best regards,
Arnd

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End of Serusers Digest, Vol 13, Issue 2