[Freeswitch-users] Sofia - calling through gateway - duplicate RTP port in SDP
kokoska.rokoska at post.cz
Sun Apr 20 08:39:21 PDT 2008
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?
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
> 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
> 4. Than UA at port 5065 try to establish media flow (183,200) to local
> user, but offers the same IP/port in SDP - 22.214.171.124:1234.
> 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
>> Provide the sip trace and the console log on pastebin and reply with
>> the urls.
> Here is pastebin url:
> Best regards,
>> 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
>>> 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
>>> 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
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>> Brian West
>> sip:brian at freeswitch.org
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
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