Hi Jan,
I tried to modify call.cpp (see Tim Flowers' e-mail below). It worked for me.
Bye.
_A_
--------------- FROM SIPP-USERS MAILING LIST ---------------------
[Sipp-users] pcapplay remote audio port truncated digits
From: Tim Flowers <[EMAIL PROTECTED]> - 2008-07-30 16:25
It looks like there is an error in parsing the remote media port from
the SDP in the 3.1 code (not sure if there is anything newer than that
where this has been corrected). It appears that the
get_remote_port_media function was truncating ports with 5 digits.
I found and corrected the problem in call.cpp, and it seems to work
fine. Here is what I changed if anyone is interested...
--- sipp.svn/call.cpp 2008-04-09 07:50:37.000000000 -0700
+++ sipp.new/call.cpp 2008-07-30 09:12:26.000000000 -0700
@@ -192,7 +192,7 @@
/* m=audio not found */
return 0;
}
- begin += strlen(pattern) - 1;
+ begin += strlen(pattern);
end = strstr(begin, "\r\n");
if (!end)
ERROR("get_remote_port_media: no CRLF found");
Please let me know if I am missing something.
Thanks,
Tim
------------------------------------------------------------
Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di John Rodriguez
Inviato: sabato 6 settembre 2008 19.15
A: Nicholas SHI; Jan Rudinský
Cc: [email protected]
Oggetto: Re: [Sipp-users] sipp remote (RTP) port handling
Is this the same issue where the incoming remote port was being truncated if >
5 characters? Isn't this fixed in the latest trunk?
> Date: Sat, 6 Sep 2008 08:57:26 +0800
> From: [EMAIL PROTECTED]
> To: [EMAIL PROTECTED]
> CC: [email protected]
> Subject: Re: [Sipp-users] sipp remote (RTP) port handling
>
> Hi Jan,
>
> My first feeling is that you touched something restricted when sending
> RTP. I mean there might be a upper port number limitation in the code
> to send RTP.
>
> Sorry. I don't look into source file. Perhaps you could modify media
> port to a lower port first at incoming 200 OK and see what's going on.
> Hopefully we can work it around before getting know what's the inside
> problem.
>
> Hope this helps.
>
> Thanks,
> Nicholas
>
> 2008/9/4 Jan Rudinský <[EMAIL PROTECTED]>:
> >
> > Hi,
> > I'm using SIPp to generate a call with RTP media. Media are sent to remote
> > side, recorded and sent back.
> > However SIPp sends media to a different remote port than offered by the
> > remote side.
> >
> > SIPp: SIP INVITE with SDP m=audio 6000 RTP/AVP 0
> > Remote: 200 OK with SDP m=audio 18436 RTP/AVP 0 101
> > SIP: RTP incoming on 6000(OK)
> > Remote: RTP incoming on 1843(instead of 18436)
> >
> > Attached: scenario graph, packet capture
> >
> > Does anyone know the solution?
> >
> > Thank you,
> >
> > JaR
> >
> >
> > --
> > Ing. Jan Rudinsky
> > Czech Technical University in Prague
> > Cesnet z.s.p.o.
> > R&D Centre (RDC) for Mobile Applications
> > [EMAIL PROTECTED]
> >
> >
> >
> >
> > -------------------------------------------------------------------------
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> > _______________________________________________
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> >
> >
>
>
>
> --
> Nicholas SHI
> -------------------------------------------------
> Tel: +86 131 5638 9168
> Email: [EMAIL PROTECTED]
> Location: Qingdao, China
> URL: http://picasaweb.google.com/shixiaomu
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