[Asterisk-Users] Strange code in rtp.c / disconnect - maybe reinvite problems

[Martin Pycko]

OK, that was obviously a 'typo' ... It's fixed.

Martin

On Tue, 25 Nov 2003, Detlef Wengorz wrote:

> Daniel Chabrol wrote:
> >
> > Hi List!
> >
> > I get "WARNING[14351]: File rtp.c, Line 1202 (ast_rtp_bridge): codec0 =
> > 524300 is not codec1 = 524300, can't do reinvite" at my asterisk console.
> >
> > The code there looks realy strange:
> >
> > codec0 = pr0->get_codec(c0);
> > codec1 = pr1->get_codec(c1);
> > ast_log(LOG_WARNING, "codec0 = %d is not codec1 = %d, can't do
> > reinvite\n",codec0,codec1);
> > /* Hey, we can't do reinvite if both parties speak diffrent codecs */
> > if (codec0 != codec1)
> > return -2;
> >
> > I think the message should only occur *after* checking equality:
> >
> > if (codec0 != codec1) {
> > ast_log(LOG_WARNING, "codec0 = %d is not codec1 = %d, can't do
> > reinvite\n",codec0,codec1);
> > return -2;
> > }
> >
> > I hoped this "can't do reinvite" would explain my disconnects from the
> > nikotel.com sip server after 60 seconds. But this little bug seems only
> > to be display-specific and not affect funtion. But maybe i oversight
>
> That's correct :-(
> but change the code like this
>
> if (codec0 != codec1) {
>      ast_log(LOG_WARNING,
>      "codec0 = %d is not codec1 = %d, can't do
> reinvite\n",codec0,codec1);
>
>      ast_mutex_unlock(&c0->lock); // unlock before return
>      ast_mutex_unlock(&c1->lock); // unlock before return
>      return -2;
> }
>
> and try again.
> maybe it helps.
>
>
>
>
> > something which still disables the reinvite even if i use
> > canreinvite=yes in my sip.conf:
> >
> > [nikotel]
> > type=friend
> > username=USERID
> > fromuser=USERID
> > secret=PASSWORD
> > host=calamar0.nikotel.com
> > canreinvite=yes
> > context=internal
> > ; no nat entry because im not using nat!
> >
> > Is there someone which is able to use Nikotel.com with the current
> > CVS-Version (in my case CVS-11/24/03-19:24:22). BTW: 0.5.0 don't work
> > too in my case (at least not longer than 60 seconds). Pulver.com calls
> > and so on are no problem. Any suggestions?
> >
> > Best regards,
> > Daniel
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> Best regards
> Detlef Wengorz <detlef at abcbtx.de>
> Detlef Wengorz <detlefw at isdn4linux.de>
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>