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

kokoska rokoska kokoska.rokoska at post.cz
Mon Apr 21 10:08:27 PDT 2008



Anthony Minessale napsal(a):
> 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.

I made the same call I decribed in my previous e-mail, but now I see 
pastebin accepts only few lines (about 600) - not all of them, thou you 
can't see mentioned call debug :-)

I grab new FreeSWITCH debug and attached it to this e-mail, beacuse I 
didn't find the way how to copy full debug to pastebin...

> 
> Also see line 38 of your pastebin there appear to be some xml errs in 
> the data you are returning from xml-curl
> 

Thank you very much, Anthony, for that point!
This happens on Directory post without "name" and "ip". I don't know 
what is this post for (nothing found in wiki) and forget to reply with 
something relevant. Now I reply with 404.

BTW: Could you (or anybody else) tell me what answer FreeSWITCH expects 
to the post above? It is on line 35 in debug...


Best regards,

kokoska.rokoska


> 
> On Mon, Apr 21, 2008 at 10:43 AM, kokoska rokoska 
> <kokoska.rokoska at post.cz <mailto: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>
>     <mailto: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>>
>      >     <mailto: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>
>     <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>
>      >      >     <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
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>      >      >      >>>
>      >      >
>      >    
>     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>
>      >     <mailto:sip%3Abrian at freeswitch.org
>     <mailto:sip%253Abrian at freeswitch.org>>
>      >     <mailto:sip%3Abrian at freeswitch.org
>     <mailto:sip%253Abrian at freeswitch.org>
>      >     <mailto:sip%253Abrian at freeswitch.org
>     <mailto:sip%25253Abrian 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:sip%3A888 at conference.freeswitch.org
>     <mailto:sip%253A888 at conference.freeswitch.org>>
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>      >     <mailto:sip%253A888 at conference.freeswitch.org
>     <mailto:sip%25253A888 at conference.freeswitch.org>>>
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>     <mailto:googletalk%25253Aconf%25252B888 at conference.freeswitch.org>>>
>      >      > pstn:213-799-1400
>      >      >
>      >      >
>      >      >
>      >    
>     ------------------------------------------------------------------------
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>      >
>      > --
>      > Anthony Minessale II
>      >
>      > FreeSWITCH http://www.freeswitch.org/
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>      >
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>      >
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> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
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