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

[kokoska rokoska]

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.

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.

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
>>
>>
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> 
> Brian West
> sip:brian at freeswitch.org
> 
> 
> 
> 
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