do you have bypass_media=true set or the inbound-bypass-media sip profile option set?<br>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<br>establish direct communication between caller and gateway.<br>
<br>Since caller has natted IP this may not work so well.<br><br>If you are setting bypass_media=true, either remove it or use proxy_media=true instead.<br><br><div class="gmail_quote">On Sun, Apr 20, 2008 at 10:39 AM, kokoska rokoska <<a href="mailto:kokoska.rokoska@post.cz">kokoska.rokoska@post.cz</a>> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><br>
My be i have a clue :-)<br>
<br>
May be FreeSWITCH thinks that offers different IPs because I set<br>
"ext-rtp-ip" to public IP of my network in SIP profile gateway is in -<br>
FreeSWITCH runs on devel machine behind NAT...<br>
<br>
Do you think it could be the reason? And if yes, how to avoid it?<br>
<br>
Best regards,<br>
<br>
kokoska.rokoska<br>
<font color="#888888"><br>
<br>
kokoska rokoska napsal(a):<br>
</font><div><div></div><div class="Wj3C7c">><br>
> Brian West napsal(a):<br>
>> Can you provide a bit more detail? I can only guess what or how<br>
>> you're trying to use it.<br>
>><br>
><br>
> Well, here is what I try to do:<br>
><br>
> 1. I have two users defined in directory which are registered with<br>
> FreeSWITCH. Both are in same SIP profile "default" (FS UA runs at port<br>
> 5065).<br>
><br>
> 2. I have different SIP profile (FS UA runs at port 7001) in which I<br>
> have defiend gateway to my telco-provider.<br>
><br>
> 3. A make a call from one of users (see 1.) to some PSTN number.<br>
> FreeSWITCH (UA at port 5065) receives INVITE, ask about credentials etc.<br>
> FreeSWITCH (UA at port 7001) successfuly authorizes against my telco<br>
> etc. and establish media flow (183,200) from local port, say 1234, IP<br>
> <a href="http://1.2.3.4" target="_blank">1.2.3.4</a>.<br>
><br>
> 4. Than UA at port 5065 try to establish media flow (183,200) to local<br>
> user, but offers the same IP/port in SDP - <a href="http://1.2.3.4:1234" target="_blank">1.2.3.4:1234</a>.<br>
><br>
> 5. Than both legs (my telco and local user) send RTP to same IP:port.<br>
> IMO it can't work. And, like I think, it don't - both legs hear nothing<br>
> and FreeSWITCH kills call after a while with cause: MEDIA_TIMEOUT.<br>
><br>
><br>
>> How are you trying to call a registered user?<br>
><br>
> I'm not sure what you are asking. I call localy registred user by<br>
> <action application='bridge' data='sofia/default/user%domain'/><br>
> and it works fine.<br>
><br>
> What don't work (see above) is to call out through gateway:<br>
> <action application='bridge' data='sofia/gateway/gw_name/number'/><br>
><br>
>> It's perfectly OK to<br>
>> receive media from two different IP's on the same port<br>
><br>
> Yes, but there is same IP and port...<br>
><br>
>> but I smell a<br>
>> bug but more so a usage case we haven't tested.<br>
>><br>
><br>
> Not sure, but it looks like it. Or I miss something important in config<br>
> options.<br>
><br>
>> Provide the sip trace and the console log on pastebin and reply with<br>
>> the urls.<br>
>><br>
><br>
> Here is pastebin url:<br>
> <a href="http://pastebin.freeswitch.org/4273" target="_blank">http://pastebin.freeswitch.org/4273</a><br>
><br>
><br>
> Best regards,<br>
><br>
> kokoska.rokoska<br>
><br>
>> /b<br>
>><br>
>> On Apr 20, 2008, at 5:42 AM, kokoska rokoska wrote:<br>
>><br>
>>> Hi all,<br>
>>><br>
>>> I have came into troubles with calling from localy registered user<br>
>>> through Sofia gateway:<br>
>>><br>
>>> FreeSWITCH offers the same RTP port in SDP to both call legs and thus<br>
>>> I'm without audio and FreeSWITCH kills the call after a while with<br>
>>> cause<br>
>>> MEDIA_TIMEOUT (this is not really valid cause, because media goes to<br>
>>> FreeSWITCH from both legs, but to the same port :-)<br>
>>><br>
>>> When I make call between localy registered users, everything looks<br>
>>> good.<br>
>>><br>
>>> I really don't know what I'm doing wrong, so any clue is very<br>
>>> appreciated :-)<br>
>>><br>
>>> BTW: I have both FreeSWITCH console log and pcap dump if someone<br>
>>> interested. And, of course, can supply any other required informations<br>
>>> about calls.<br>
>>><br>
>>> Thanks in advance, best regards<br>
>>><br>
>>> kokoska.rokoska<br>
>>><br>
>>><br>
>>> _______________________________________________<br>
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>> Brian West<br>
>> <a href="mailto:sip%3Abrian@freeswitch.org">sip:brian@freeswitch.org</a><br>
>><br>
>><br>
>><br>
>><br>
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>><br>
><br>
><br>
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