[Freeswitch-users] Sofia - calling through gateway - duplicate RTP port in SDP
kokoska rokoska
kokoska.rokoska at post.cz
Mon Apr 21 08:43:35 PDT 2008
Anthony Minessale napsal(a):
> please set the environment variable TPORT_LOG=1 to log the sip traffic
> inline.
> then capture all of the console output reproduce your issue
> before you try the call, please enter the command "console loglevel
> debug" at the CLI
>
> also, just to be safe, examine your dialplan for any instances of
> bypass_media and examine all of your sip profiles for the
> inbound-bypass-media param.
Many thanks, Anthony, for your help!
I double-checked my sip profiles and dialplan and found nothing which
looks like "bypass"...
Console log I have submited to pastebin:
http://pastebin.freeswitch.org/4277
Best regards,
kokoska.rokoska
>
>
> On Mon, Apr 21, 2008 at 8:52 AM, kokoska rokoska
> <kokoska.rokoska at post.cz <mailto:kokoska.rokoska at 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 at post.cz <mailto:kokoska.rokoska at post.cz>
> <mailto:kokoska.rokoska at post.cz <mailto:kokoska.rokoska at 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> <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>
> > <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 at lists.freeswitch.org
> <mailto:Freeswitch-users at lists.freeswitch.org>
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> > >>>
> >
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> > >>> http://www.freeswitch.org
> > >> Brian West
> > >> sip:brian at freeswitch.org
> <mailto:sip%3Abrian at freeswitch.org>
> <mailto:sip%3Abrian at freeswitch.org
> <mailto:sip%253Abrian at freeswitch.org>>
> > >>
> > >>
> > >>
> > >>
> > >> _______________________________________________
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> > >>
> >
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> > >>
> > >
> > >
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> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com
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> > <mailto:PAYPAL%3Aanthony.minessale at gmail.com
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> > IRC: irc.freenode.net <http://irc.freenode.net>
> <http://irc.freenode.net> #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org
> <mailto:sip%3A888 at conference.freeswitch.org>
> > <mailto:sip%3A888 at conference.freeswitch.org
> <mailto:sip%253A888 at conference.freeswitch.org>>
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> <http://iax:guest@conference.freeswitch.org/888>
> > <http://iax:guest@conference.freeswitch.org/888>
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> > pstn:213-799-1400
> >
> >
> >
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>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
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>
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