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:
|Time | 192.168.0.3 | 192.168.0.2 |
|42.043 | INVITE SDP ( g711A) |SIP From:
sip:[EMAIL PROTECTED]:5060 To:sip:[EMAIL PROTECTED]:5060
| |(5060) ------------------> (5060) |
|42.044 | 180 Ringing |SIP Status
| |(5060) <------------------ (5060) |
|42.044 | 200 OK SDP ( g711U) |SIP Status
| |(5060) <------------------ (5060) |
|42.045 | ACK | |SIP Request
| |(5060) ------------------> (5060) |
|42.045 | RTP (g711A) |RTP Num
packets:1 Duration:0.000s SSRC:0x708EBBB9
| |(6016) ------------------> (6000) |
|42.046 | RTP (g711A) |RTP Num
packets:362 Duration:7.219s SSRC:0x708EBBB9
| |(6016) <------------------ (6000) |
|62.049 | RTP (telephone-event) DTMF One 1 |RTP Num packets:8
Duration:0.139s SSRC:0xE05384E
| |(6016) <------------------ (6000) |
|63.050 | BYE | |SIP Request
| |(5060) ------------------> (5060) |
|63.051 | 200 OK | |SIP Status
| |(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:
|Time | 192.168.0.3 | 192.168.0.2 |
|0.000 | INVITE SDP ( g711A) |SIP From:
sip:[EMAIL PROTECTED]:5060 To:sip:[EMAIL PROTECTED]:5060
| |(5060) ------------------> (5060) |
|0.000 | 180 Ringing |SIP Status
| |(5060) <------------------ (5060) |
|0.001 | 200 OK SDP ( g711U) |SIP Status
| |(5060) <------------------ (5060) |
|0.002 | ACK | |SIP
Request
| |(5060) ------------------> (5060) |
|0.006 | RTP (g711A) |RTP Num
packets:362 Duration:7.219s SSRC:0x708EBBB9
| |(6000) <------------------ (6000) |
|20.013 | RTP (telephone-event) DTMF One 1 |RTP Num packets:8
Duration:0.140s SSRC:0xE05384E
| |(6000) <------------------ (6000) |
|21.024 | BYE | |SIP Request
| |(5060) ------------------> (5060) |
|21.025 | 200 OK | |SIP Status
| |(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:
|Time | 192.168.0.3 | 192.168.0.2 |
|21.027 | INVITE SDP ( g711A) |SIP From:
sip:[EMAIL PROTECTED]:5060 To:sip:[EMAIL PROTECTED]:5060
| |(5060) ------------------> (5060) |
|21.028 | 180 Ringing |SIP Status
| |(5060) <------------------ (5060) |
|21.028 | 200 OK SDP ( g711U) |SIP Status
| |(5060) <------------------ (5060) |
|21.029 | RTP (g711A) |RTP Num
packets:724 Duration:7.229s SSRC:0x708EBBB9
| |(6008) ------------------> (6000) |
|21.031 | ACK | |SIP
Request
| |(5060) ------------------> (5060) |
|21.036 | RTP (g711A) |RTP Num
packets:362 Duration:7.222s SSRC:0x708EBBB9
| |(6012) <------------------ (6000) |
|41.032 | RTP (101) DTMF One 1 |RTP Num packets:15
Duration:0.139s SSRC:0xE05384E
| |(6008) ------------------> (6000) |
|41.042 | RTP (telephone-event) DTMF One 1 |RTP Num packets:8
Duration:0.139s SSRC:0xE05384E
| |(6012) <------------------ (6000) |
|42.051 | BYE | |SIP
Request
| |(5060) ------------------> (5060) |
|42.052 | 200 OK | |SIP Status
| |(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
-------------------------------------------------------------------------
This SF.Net email is sponsored by the Moblin Your Move Developer's challenge
Build the coolest Linux based applications with Moblin SDK & win great prizes
Grand prize is a trip for two to an Open Source event anywhere in the world
http://moblin-contest.org/redirect.php?banner_id=100&url=/
_______________________________________________
Sipp-users mailing list
[email protected]
https://lists.sourceforge.net/lists/listinfo/sipp-users
