I believe the call flow in Wireshark for simultaneous calls is not displaying
the correct information due to the RTP sequence number being the same for all
simultaneous calls. For example if '-users 50' option is being used, all 50
simultaneous calls will use the same RTP sequence number, therefore not
allowing Wireshark to detect the correct call flow.
Is this possible to get a different RTP sequence number for all simultaneous
calls created with the '-users' options?
Patrick, Olivier,
I noticed you were both involved in the following request;
http://sourceforge.net/tracker/?func=detail&atid=637567&aid=1759105&group_id=104305,
could you let me know the outcome of this request?
Thanks,
Antoine
----- Forwarded Message ----
From: Antoine <[EMAIL PROTECTED]>
To: [email protected]
Sent: Wednesday, 26 November, 2008 0:30:56
Subject: Re: [Sipp-users] Inconsistent call flows in Wireshark using -RTP_ECHO
and -users 2 (or higher than 2)
After further investigation, the call flows in Wireshark (issued from
'Statistics' menu, 'VoIP Calls', then select one of the calls and click the
'Graph' button) not being shown properly seems to be due to Wireshark itself.
According to the packets, the calls generated by SIPp and -RTP_ECHO seems to be
perfectly formed.
In the latest network trace which I've gathered only 2 calls were created at
the same time, if I filter only the packets from the first call then the call
flow in Wireshark is being shown properly.
I did not notice any issues if I only generate one call at a time (i.e. using
'-users 1' and '-RTP_ECHO' options). Basically the call flow issue only occurs
in Wireshark for calls created with options -RTP_ECHO & -users X (with X higher
or equal to 2).
The only workaround I found is to use the 'Flow Graph...' option from the
'Statistics' menu, however it is a bit messy as it shows the call flow for all
calls at the same time.
It would be nice if someone could backup what I am saying before I make the
Wireshark guys aware.
Antoine
----- Forwarded Message ----
From: Antoine <[EMAIL PROTECTED]>
To: [email protected]
Sent: Monday, 24 November, 2008 18:44:50
Subject: [Sipp-users] Inconsistent call flows using RTP_ECHO - bug?
I am creating 2 G711alaw bidirectional calls between a UAC and UAS using the
following commands:
UAC (IP 192.168.0.3): ./sipp -sf G711alaw.xml 192.168.0.2 -users 2 -mi
192.168.0.3 -i 192.168.0.3
UAS (IP 192.168.0.2): ./sipp -sn uas -mi 192.168.0.2 -i 192.168.0.2
-rtp_echo
I am using the -rtp_echo setting on the UAS to return the G711a pcap file back
to the UAC. It works pretty well most of the time, the call flow for a valid
call looks like that:
192.168.0.3 192.168.0.2
INVITE SDP ( g711A)
(5060) ------------------> (5060)
180 Ringing
(5060) <------------------ (5060)
200 OK SDP ( g711U)
(5060) <------------------ (5060)
ACK
(5060) ------------------> (5060)
RTP (g711A)
(6016) ------------------> (6000)
RTP (g711A)
(6016) <------------------ (6000)
RTP (telephone-event) DTMF One 1
(6016) <------------------ (6000)
BYE
(5060) ------------------> (5060)
200 OK
(5060) <------------------ (5060)
However I found some issues after collecting a network trace on the UAC using
Wireshark. Some of the calls, which appear as COMPLETED in Wireshark, only
contain the pcap file going from the UAS back to the UAC. Call flow in this
case is as follows:
192.168.0.3 192.168.0.2
INVITE SDP ( g711A)
(5060) ------------------> (5060)
180 Ringing
(5060) <------------------ (5060)
200 OK SDP ( g711U)
(5060) <------------------ (5060)
ACK
(5060) ------------------> (5060)
RTP (g711A)
(6000) <------------------ (6000)
RTP (telephone-event) DTMF One 1
(6000) <------------------ (6000)
BYE
(5060) ------------------> (5060)
200 OK
(5060) <------------------ (5060)
Other calls show the SIP Request (ACK) occurring between the first and the
second G711a pcap file instead of occuring before the first pcap file, i.e:
192.168.0.3 192.168.0.2
INVITE SDP ( g711A)
(5060) ------------------> (5060)
180 Ringing
(5060) <------------------ (5060)
200 OK SDP ( g711U)
(5060) <------------------ (5060)
RTP (g711A)
(6008) ------------------> (6000)
ACK
(5060) ------------------> (5060)
RTP (g711A)
(6012) <------------------ (6000)
RTP (101) DTMF One 1
(6008) ------------------> (6000)
RTP (telephone-event) DTMF One 1
(6012) <------------------ (6000)
BYE
(5060) ------------------> (5060)
200 OK
(5060) <------------------ (5060)
It looks like despite SIPp and Wireshark showing the calls as sucessful, some
of the calls are not be trully valid as explained above.
-> How can I make sure only the calls with a valid call flow (1st call flow
above) are being detected as successful by SIPp?
The log files are available through one of my previous post.
Thanks,
Antoine
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