Olivier,

Thank you for your quick reply.

With -i, the INVITE carries correct SDP with c=IN IP4 192.168.53.61. But the
problem stays still. In my opinion, sipp should echo all rtp/udp packets on
specific port. in the tcpdump, The rtp arrives at server side with correct
Sourse and Dest, sipp could echo them as it knows enough info.

Have you any suggestion else?

Best regards
Zhou

On 2/2/07, Olivier Jacques <[EMAIL PROTECTED]> wrote:

The INVITE sent by the proxy (192.168.52.3) to SIPp UAS contains an SDP
with c=IN IP4 127.0.0.1.
So SIPp UAS will send the RTP back to 127.0.0.1. That's why you don't
see packets.

Please try adding "-i" on the client side, I guess this is why you get a
"127.0.0.1" in the SDP.

Olivier.

sip zhou wrote:
> Hi Olivier,
>
> seems still no way to echo the rtp with the -mi option. any other
> options should I try out?
> anyway, thanks for your advice.
>
> regards!
> Zhou
>
> On 2/2/07, *Olivier Jacques* <[EMAIL PROTECTED]
> <mailto:[EMAIL PROTECTED]>> wrote:
>
>     Hello,
>
>     on server side, try adding "-mi 192.168.48.161
>     <http://192.168.48.161> <http://192.168.48.161>".
>
>     Olivier.
>     sip zhou wrote:
>     > Dear sipp team,
>     >
>     > I tried to play rtp echo by following commands, but failed. it
>     seemed
>     > like rtp was well played by sipp client and was reached at sipp
>     server
>     > side. but no echo by sipp server.
>     >
>     > client: on 192.168.53.61 <http://192.168.53.61>
>     <http://192.168.53.61>
>     > ./sipp sip_proxy_server -sf media.xml -inf uac_info.CSV -mp 6000
-mi
>     > 192.168.53.61 <http://192.168.53.61> <http://192.168.53.61> -r 1
>     -m 1
>     >
>     > server: on 192.168.48.161 <http://192.168.48.161>
>     <http://192.168.48.161 <http://192.168.48.161>>
>     > ./sipp sip_proxy_server -sf call_uas.xml -mp 6000 -rtp_echo
>     >
>     > sip_proxy_server: 192.168.52.3 <http://192.168.52.3>
>     <http://192.168.52.3>
>     >
>     > I used the latest snapshot in your download page.
>     > I attach you some tcpdumps, Please take a look at them if
possible.
>     >
>     > Thanks very much!
>     > Zhou
>     >
>     > attached is the tcpdump at sipp server side.
>     >
>     >
>     > ------------------------
>     > the client scenario:
>     >
>     > ------------------------------ Scenario Screen -------- [1-9]:
>     Change
>     > Screen --
>     >   Call-rate(length)     Port
Total-time  Total-calls  Remote-host
>     >    1.0(0 ms)/1.000s   5060      61.32 s            1
>     > 192.168.52.3:5060(UDP)
>     >
>     >   Call limit reached (-m 1), 0.314 s period  3 ms scheduler
>     resolution
>     >   0 calls (limit 180)                    Peak was 1 calls, after 1
s
>     >   0 Running, 1 Paused, 0 Woken up
>     >   0 out-of-call msg (discarded)
>     >   1 open sockets
>     >   2951 Total RTP pckts sent              8.240 last period RTP
>     rate (kB/s)
>     >
>     >                                  Messages  Retrans   Timeout
>     > Unexpected-Msg
>     >       INVITE ---------->         1         0         0
>     >          100 <----------         1         0                   0
>     >          180 <----------         0         0                   0
>     >          401 <----------         1         0                   0
>     >          ACK ---------->         1         0
>     >
>     >       INVITE ---------->         1         0         0
>     >          100 <----------         1         0                   0
>     >          180 <----------         1         0                   0
>     >          200 <----------  E-RTD1 1         0                   0
>     >          ACK ---------->         1         0
>     >
>     >               [ NOP ]
>     >        Pause [    60.0s]         1                             0
>     >        Pause [      0ms]         1                             0
>     >          BYE ---------->         1         0         0
>     >          200 <----------         1         0                   0
>     >
>     > ------------------------------ Test Terminated
>     > --------------------------------
>     >
>     >
>     > the server scenario:
>     >
>     >  Port   Total-time  Total-calls  Transport
>     >   5060      82.05 s            1  UDP
>     >
>     >   0 new calls during 1.001 s period      2 ms scheduler resolution
>     >   0 calls                                Peak was 1 calls, after 7
s
>     >   0 Running, 0 Paused, 0 Woken up
>     >   1 open sockets
>     >   0 Total echo RTP pckts 1st stream      0.000 last period RTP
>     rate (kB/s)
>     >   0 Total echo RTP pckts 2nd stream       0.000 last period RTP
>     rate (kB/s)
>     >
>     >                                  Messages  Retrans   Timeout
>     > Unexpected-Msg
>     >   ----------> INVITE             1         0                   0
>     >   <---------- 180                1         0
>     >   <---------- 200                1         0         0
>     >   ----------> ACK         E-RTD1 1         0                   0
>     >
>     >   ----------> BYE                1         0                   0
>     >   <---------- 200                1         0
>     > ------------------------------ Sipp Server Mode
>     > -------------------------------
>     >
>     >
>     >
>     >
>     >
>     >
>     >
>
------------------------------------------------------------------------
>
>     >
>     >
>
-------------------------------------------------------------------------
>     > Using Tomcat but need to do more? Need to support web services,
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>     Apache Geronimo
>     >
>
http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642
>     <
http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642>
>     >
>
------------------------------------------------------------------------
>     >
>     > _______________________________________________
>     > Sipp-users mailing list
>     > [email protected]
>     <mailto:[email protected]>
>     > https://lists.sourceforge.net/lists/listinfo/sipp-users
>     >
>
>
>     --
>     Olivier
>     HP OpenCall Software
>     http://www.hp.com/go/opencall/
>
>


--
Olivier
HP OpenCall Software
http://www.hp.com/go/opencall/


-------------------------------------------------------------------------
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