[Freeswitch-users] Sofia - calling through gateway - duplicate RTP port in SDP
Anthony Minessale
anthony.minessale at gmail.com
Mon Apr 21 09:10:14 PDT 2008
Please do the same trace but actually make the phone call that you were
describing in your original email so i can see the output of the FS console
and the SIP trace together.
Also see line 38 of your pastebin there appear to be some xml errs in the
data you are returning from xml-curl
On Mon, Apr 21, 2008 at 10:43 AM, kokoska rokoska <kokoska.rokoska at post.cz>
wrote:
>
>
>
> 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>
> > > <mailto:Freeswitch-users at lists.freeswitch.org
> > <mailto:Freeswitch-users at lists.freeswitch.org>>
> > > >>>
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > >>>
> > >
> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > >>> http://www.freeswitch.org
> > > >> Brian West
> > > >> sip:brian at freeswitch.org <sip%3Abrian at freeswitch.org>
> > <mailto:sip%3Abrian at freeswitch.org <sip%253Abrian at freeswitch.org>>
> > <mailto:sip%3Abrian at freeswitch.org <sip%253Abrian at freeswitch.org>
> > <mailto:sip%253Abrian at freeswitch.org<sip%25253Abrian at freeswitch.org>
> >>
> > > >>
> > > >>
> > > >>
> > > >>
> > > >> _______________________________________________
> > > >> Freeswitch-users mailing list
> > > >> Freeswitch-users at lists.freeswitch.org
> > <mailto:Freeswitch-users at lists.freeswitch.org>
> > > <mailto:Freeswitch-users at lists.freeswitch.org
> > <mailto:Freeswitch-users at lists.freeswitch.org>>
> > > >>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > >>
> > >
> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > >> http://www.freeswitch.org
> > > >>
> > > >
> > > >
> > > > _______________________________________________
> > > > Freeswitch-users mailing list
> > > > Freeswitch-users at lists.freeswitch.org
> > <mailto:Freeswitch-users at lists.freeswitch.org>
> > > <mailto:Freeswitch-users at lists.freeswitch.org
> > <mailto:Freeswitch-users at lists.freeswitch.org>>
> > > >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > >
> > >
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> > > > http://www.freeswitch.org
> > > >
> > >
> > >
> > > _______________________________________________
> > > Freeswitch-users mailing list
> > > Freeswitch-users at lists.freeswitch.org
> > <mailto:Freeswitch-users at lists.freeswitch.org>
> > > <mailto: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
> > >
> > >
> > >
> > >
> > > --
> > > Anthony Minessale II
> > >
> > > FreeSWITCH http://www.freeswitch.org/
> > > ClueCon http://www.cluecon.com/
> > >
> > > AIM: anthm
> > > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> > <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
> > > <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> > <mailto:MSN%253Aanthony_minessale at hotmail.com<MSN%25253Aanthony_minessale at hotmail.com>
> >>
> > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
> > > <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> > <mailto:PAYPAL%253Aanthony.minessale at gmail.com<PAYPAL%25253Aanthony.minessale at gmail.com>
> >>
> > > IRC: irc.freenode.net <http://irc.freenode.net>
> > <http://irc.freenode.net> #freeswitch
> > >
> > > FreeSWITCH Developer Conference
> > > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> > <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> > > <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> > <mailto:sip%253A888 at conference.freeswitch.org<sip%25253A888 at conference.freeswitch.org>
> >>
> > > iax:guest at conference.freeswitch.org/888
> > <http://iax:guest@conference.freeswitch.org/888>
> > > <http://iax:guest@conference.freeswitch.org/888>
> > > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> > > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> > <mailto:googletalk%253Aconf%252B888 at conference.freeswitch.org<googletalk%25253Aconf%25252B888 at conference.freeswitch.org>
> >>
> > > pstn:213-799-1400
> > >
> > >
> > >
> >
> ------------------------------------------------------------------------
> > >
> > > _______________________________________________
> > > Freeswitch-users mailing list
> > > Freeswitch-users at lists.freeswitch.org
> > <mailto:Freeswitch-users at lists.freeswitch.org>
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >
> > UNSUBSCRIBE:
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> > > http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > <mailto:Freeswitch-users at lists.freeswitch.org>
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
> > IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> > iax:guest at conference.freeswitch.org/888
> > <http://iax:guest@conference.freeswitch.org/888>
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> > pstn:213-799-1400
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
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> http://www.freeswitch.org
>
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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