Hi,
YES no it works, with snapshot 2006-08-16.
Thanks, this is a great tool!!
BR Ulf
From: Olivier Jacques
[mailto:[EMAIL PROTECTED]
Sent: den 18 augusti 2006 12:21
To: Ulf Wätterstam
Cc: guillaume teissier;
[email protected]
Subject: Re: [Sipp-users] Two
calls in same scenario with pcap play
Hi Ulf,
yes, I would recommend to upgrade to use the latest snapshot (http://sipp.sourceforge.net/snapshots/)
and try again.
Some things have changed around SDP parsing since 1.1rc5.
Note that you will have to update your scenario and replace <exec>
actions with play_pcap_audio (see embedded uac_pcap scenario).
Let us know if this is not enough.
Olivier.
On 8/18/06, Ulf
Wätterstam <[EMAIL PROTECTED]>
wrote:
After big hazel with
tcpdump (it cutes the SIP messages) I used the -trace_msg flag in SIPp to check
the SDP values. I think this is what's happening.
In first Call, we use RTP port 10000 and 11924. This is correct according to
SDP values.
In Second Call, we should use RTP port 10000 and 18422, according to SDP
values. But sipp still send the RTP message to port 11924.
I use sipp 1.1rc5. Does anyone know if this works in any later release or if
there is a workaround?
BR Ulf W
-----Original Message-----
From: guillaume teissier [mailto:[EMAIL PROTECTED]]
Sent: den 16 augusti 2006 19:17
To: Ulf Wätterstam
Cc: [email protected]
Subject: Re: [Sipp-users] Two calls in same scenario with pcap play
Hi Ulf,
Can you take a network capture of the whole exchange? One thing u can
check is answered SDP against RTP ports used by SIPP.
There may also be a sequence trouble as RTP packets emitted for the
second time will have the same TS and SEQ numbers as first ones.
BR
Guillaume
2006/8/16, Ulf Wätterstam <[EMAIL PROTECTED]>:
>
>
>
> Hi,
>
>
>
> I would like to make two calls in the same scenario and play a RTP stream.
I
> took the built in scenario uac_pcap and simply duplicated the call part.
>
> To verify I used a soft phone (X-lite) as the receiver of the call. In the
> first call (1-8) everything works fine, but in the second call the voice
> (RTP stream 15,16) does not reach the soft phone. I do not understand what
I
> do wrong, can anyone help me?
>
>
>
> SIPp
UAC
Remote
>
> |(1) INVITE
|
>
> |------------------>|
>
> |(2) 100 (optional) |
>
> |<------------------|
>
> |(3) 180 (optional) |
>
> |<------------------|
>
> |(4)
200 |
>
> |<------------------|
>
> |(5)
ACK |
>
> |------------------>|
>
>
| |
>
> |(6) RTP send (8s) |
>
> |============>|
>
>
|
|
>
> |(7) RTP send (8s) |
>
> |============>|
>
>
|
|
>
> |(8)
BYE |
>
> |------------------>|
>
> |(9)
200 |
>
> |<------------------|
>
> Pause (10s)
>
> |(10)
INVITE |
>
> |------------------>|
>
> |(11) 100 (optional) |
>
> |<------------------|
>
> |(12) 180 (optional) |
>
> |<------------------|
>
> |(13)
200 |
>
> |<------------------|
>
> |(14)
ACK |
>
> |------------------>|
>
>
| |
>
> |(15) RTP send (8s) |
>
> |============>|
>
>
|
|
>
> |(16) RTP send (8s) |
>
> |============>|
>
>
|
|
>
> |(17)
BYE
|
>
> |------------------>|
>
> |(18)
200
|
>
> |<------------------|
>
>
>
> Regards Ulf Wätterstam
>
>
> -------------------------------------------------------------------------
> Using Tomcat but need to do more? Need to support web services, security?
> Get stuff done quickly with pre-integrated technology to make your job
> easier
> Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo
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>
> _______________________________________________
> Sipp-users mailing list
> [email protected]
> https://lists.sourceforge.net/lists/listinfo/sipp-users
>
>
>
-------------------------------------------------------------------------
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|
-------------------------------------------------------------------------
Using Tomcat but need to do more? Need to support web services, security?
Get stuff done quickly with pre-integrated technology to make your job easier
Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo
http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642
_______________________________________________
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