On Mon, Apr 21, 2008 at 8:52 AM, kokoska rokoska <kokoska.rokoska@post.cz> wrote:

Thank you very much, Anthony, for your answer!

Anthony Minessale napsal(a):

> do you have bypass_media=true set or the inbound-bypass-media sip
> profile option set?

No, no. I use neither of them (if they are not default :-).

BTW: I need FreeSWITCH in media path due to four reasons:
1. To handle NATed clients.
2. To kill call without RTP.
2. To process DTMF.
3. To transcode between codecs if needed.

> This call flow looks to me like proxy mode where the sdp of the inbound
> leg is offered to the remote leg to attempt to
> establish direct communication between caller and gateway.
>

If yes it is (my?) mistake. I don't want proxy mode :-)

> Since caller has natted IP this may not work so well.
>

I think so.

> If you are setting bypass_media=true, either remove it or use
> proxy_media=true instead.
>

May be - in the future - i will utilize proxy_media=true too, but now I
need two independant call legs with two independant media streams with
FreeSWITCH between them.

If you have any suggestion what should I do to eliminate my trouble,
please give me the clue. I'm afraid I have no idea what else I could try
to do :-)

Thanks in advance, best regards

kokoska.rokoska

> On Sun, Apr 20, 2008 at 10:39 AM, kokoska rokoska

> <kokoska.rokoska@post.cz <mailto:kokoska.rokoska@post.cz>> wrote:
>
>
> My be i have a clue :-)
>
> May be FreeSWITCH thinks that offers different IPs because I set
> "ext-rtp-ip" to public IP of my network in SIP profile gateway is in -
> FreeSWITCH runs on devel machine behind NAT...
>
> Do you think it could be the reason? And if yes, how to avoid it?
>
> Best regards,
>
> kokoska.rokoska
>
>
> kokoska rokoska napsal(a):
> >
> > Brian West napsal(a):
> >> Can you provide a bit more detail? I can only guess what or how
> >> you're trying to use it.
> >>
> >
> > Well, here is what I try to do:
> >
> > 1. I have two users defined in directory which are registered with
> > FreeSWITCH. Both are in same SIP profile "default" (FS UA runs at
> port
> > 5065).
> >
> > 2. I have different SIP profile (FS UA runs at port 7001) in which I
> > have defiend gateway to my telco-provider.
> >
> > 3. A make a call from one of users (see 1.) to some PSTN number.
> > FreeSWITCH (UA at port 5065) receives INVITE, ask about
> credentials etc.
> > FreeSWITCH (UA at port 7001) successfuly authorizes against my telco
> > etc. and establish media flow (183,200) from local port, say 1234, IP

> > 1.2.3.4 <http://1.2.3.4>.

> >
> > 4. Than UA at port 5065 try to establish media flow (183,200) to
> local
> > user, but offers the same IP/port in SDP - 1.2.3.4:1234

> <http://1.2.3.4:1234>.

> >
> > 5. Than both legs (my telco and local user) send RTP to same IP:port.
> > IMO it can't work. And, like I think, it don't - both legs hear
> nothing
> > and FreeSWITCH kills call after a while with cause: MEDIA_TIMEOUT.
> >
> >
> >> How are you trying to call a registered user?
> >
> > I'm not sure what you are asking. I call localy registred user by
> > <action application='bridge' data='sofia/default/user%domain'/>
> > and it works fine.
> >
> > What don't work (see above) is to call out through gateway:
> > <action application='bridge' data='sofia/gateway/gw_name/number'/>
> >
> >> It's perfectly OK to
> >> receive media from two different IP's on the same port
> >
> > Yes, but there is same IP and port...
> >
> >> but I smell a
> >> bug but more so a usage case we haven't tested.
> >>
> >
> > Not sure, but it looks like it. Or I miss something important in
> config
> > options.
> >
> >> Provide the sip trace and the console log on pastebin and reply with
> >> the urls.
> >>
> >
> > Here is pastebin url:
> > http://pastebin.freeswitch.org/4273
> >
> >
> > Best regards,
> >
> > kokoska.rokoska
> >
> >> /b
> >>
> >> On Apr 20, 2008, at 5:42 AM, kokoska rokoska wrote:
> >>
> >>> Hi all,
> >>>
> >>> I have came into troubles with calling from localy registered user
> >>> through Sofia gateway:
> >>>
> >>> FreeSWITCH offers the same RTP port in SDP to both call legs
> and thus
> >>> I'm without audio and FreeSWITCH kills the call after a while with
> >>> cause
> >>> MEDIA_TIMEOUT (this is not really valid cause, because media
> goes to
> >>> FreeSWITCH from both legs, but to the same port :-)
> >>>
> >>> When I make call between localy registered users, everything looks
> >>> good.
> >>>
> >>> I really don't know what I'm doing wrong, so any clue is very
> >>> appreciated :-)
> >>>
> >>> BTW: I have both FreeSWITCH console log and pcap dump if someone
> >>> interested. And, of course, can supply any other required
> informations
> >>> about calls.
> >>>
> >>> Thanks in advance, best regards
> >>>
> >>> kokoska.rokoska
> >>>
> >>>
> >>> _______________________________________________
> >>> Freeswitch-users mailing list
> >>> Freeswitch-users@lists.freeswitch.org

> <mailto:Freeswitch-users@lists.freeswitch.org>

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> >>>
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> >> Brian West

> >> sip:brian@freeswitch.org <mailto:sip%3Abrian@freeswitch.org>

> >>
> >>
> >>
> >>
> >> _______________________________________________
> >> Freeswitch-users mailing list
> >> Freeswitch-users@lists.freeswitch.org

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> >>
> >
> >
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> --
> Anthony Minessale II
>
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