[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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>     <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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>      >
>      > --
>      > Anthony Minessale II
>      >
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> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
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