[Freeswitch-users] Sofia - calling through gateway - duplicate RTP port in SDP

kokoska rokoska kokoska.rokoska at post.cz
Mon Apr 21 06:52:52 PDT 2008


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>> 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 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 <mailto:sip%3Abrian at freeswitch.org>
>      >>
>      >>
>      >>
>      >>
>      >> _______________________________________________
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>      >>
>      >
>      >
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> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
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