[Serusers] ACK
Greger V. Teigre
greger at teigre.com
Thu Jul 21 07:20:56 CEST 2005
Just curious: Have you used the ONsip.org Getting Started guide, configs and
getting started source package!? Rtpproxy is included, while mediaproxy is
a standalone package where everything is prepared.
Getting this far, you can try to the udp mode:
# We set up requests over udp
modparam("nathelper", "rtpproxy_sock", "udp:10.56.0.5:22222")
Start rtpproxy this way:
rtpproxy -l your_up -s udp:*
g-)
Sebastian Kühner wrote:
> Hi,
>
> I don't have a rtpproxy.pid file. You mean the *.sock file?
>
> Here is the permission:
> srwxr-xr-x 1 root root 0 2005-07-20 18:18 rtpproxy.sock=
>
> The rtpproxy has to create a pid-file?
>
> Thanks!
>
>
> ----- Original Message -----
> From: "harry gaillac" <gaillacharry at yahoo.fr>
> To: "Sebastian Kühner" <skuehner at veraza.com>
> Sent: Wednesday, July 20, 2005 5:22 PM
> Subject: Re: [Serusers] ACK
>
>
>> did you check /var/run/*/rtpproxy.pid
>>
>> --- Sebastian Kühner <skuehner at veraza.com> a écrit :
>>
>>> Hi!
>>>
>>> Thanks for your question ;-)
>>>
>>> I'm using Slackware...
>>>
>>> ----- Original Message -----
>>> From: "harry gaillac" <gaillacharry at yahoo.fr>
>>> To: "Sebastian Kühner" <skuehner at veraza.com>
>>> Sent: Wednesday, July 20, 2005 5:07 PM
>>> Subject: Re: [Serusers] ACK
>>>
>>>
>>>> What's your distro Debian, .. ?
>>>>
>>>> --- Sebastian Kühner <skuehner at veraza.com> a écrit
>>>>
>>>>
>>>>> It should... but it doesn't. I have ser 0.9.0
>>> and
>>>>> the latest rtpproxy
>>>>> version.
>>>>>
>>>>> WARNING: rtpp_test: can't get version of the RTP
>>>>> proxy
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> ----- Original Message -----
>>>>> From: "harry gaillac" <gaillacharry at yahoo.fr>
>>>>> To: "Sebastian Kühner" <skuehner at veraza.com>
>>>>> Sent: Wednesday, July 20, 2005 1:44 PM
>>>>> Subject: Re: [Serusers] ACK
>>>>>
>>>>>
>>>>>> your rtpproxy should work !
>>>>>>
>>>>>> --- Sebastian Kühner <skuehner at veraza.com> a
>>> écrit
>>>>>>
>>>>>>
>>>>>>> Hi,
>>>>>>>
>>>>>>> Ok, my rtpproxy doesn't work, so I try it with STUN.
>>>>>>> When I look at my
>>>>>>> SIP-messages I get the information, that the
>>>>> audio
>>>>>>> stream has to go through
>>>>>>> my public IP... but I don't hear anything (I
>>>>> have
>>>>>>> the volume on maximum).
>>>>>>>
>>>>>>> The Invite comes with this message:
>>>>>>>
>>>>>>> v=0.
>>>>>>> o=- 3330865830 3330865830 IN IP4
>>>>> xxx.xxx.xxx.xxx.
>>>>>>> <-- Public IP
>>>>>>> s=SJphone.
>>>>>>> c=IN IP4 xxx.xxx.xxx.xxx
>>> <--
>>>>>>> Public IP
>>>>>>> t=0 0.
>>>>>>> a=direction:active.
>>>>>>> m=audio 16482 RTP/AVP 3 8 0 101.
>>>>>>> a=rtpmap:3 GSM/8000.
>>>>>>> a=rtpmap:8 PCMA/8000.
>>>>>>> a=rtpmap:0 PCMU/8000.
>>>>>>> a=rtpmap:101 telephone-event/8000.
>>>>>>> a=fmtp:101 0-11,16.
>>>>>>>
>>>>>>> Doesn't that mean, that the audio-stream has to go
>>>>>>> through my public IP now?
>>>>>>> Both sides doesn't hear anything...
>>>>>>>
>>>>>>> What's wrong?
>>>>>>>
>>>>>>> Sebastian
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> ----- Original Message -----
>>>>>>> From: "Greger V. Teigre" <greger at teigre.com>
>>>>>>> To: "Sebastian Kühner"
>>> <skuehner at veraza.com>;
>>>>>>> <serusers at lists.iptel.org>
>>>>>>> Sent: Wednesday, July 20, 2005 2:24 AM
>>>>>>> Subject: Re: [Serusers] ACK
>>>>>>>
>>>>>>>
>>>>>>>> Sebastian,
>>>>>>>> I know many people don't like STUN. However, I have good
>>>>>>>> experiences with STUN and prefer to use STUN as a "first layer
>>>>>>>> defence." For many NATs I then avoid the proxying. However,
>>>>>>>> there
>>> are
>>>>> some
>>>>>>> things that can go wrong:
>>>>>>>> For one, you need to make sure that the
>>> STUN
>>>>>>> server is running correctly
>>>>>>> on
>>>>>>>> two ports and two IP addresses. If you for
>>>>> example
>>>>>>> have a firewall
>>>>>>> blocking
>>>>>>>> one port, STUN will give the wrong result.
>>> But
>>>>> the
>>>>>>> biggest problem can be
>>>>>>>> faulty STUN implementations in the EUCs. They normally behave
>>>>>>>> ok for the most standard NATs, but there are some
>>>>>>> non-standard NATs and the EUC's
>>>>>>>> behavior can be unpredictable. Also, some EUCs try to rewrite
>>>>>>>> the IP:port even if they are behind a symmetric NAT
>>> (or if
>>>>> the
>>>>>>> STUN server is not
>>>>>>>> correctly set up, the EUC will conclude
>>> with
>>>>> the
>>>>>>> wrong result).
>>>>>>>> If you know the clients you are going
>>> to
>>>>> use,
>>>>>>> you can test and limit
>>>>>>> the
>>>>>>>> problems and STUN can be a great cost
>>> saver!
>>>>> If
>>>>>>> your gateway supports
>>>>>>>> active media (direction=active), then you only have IP-2-IP
>>>>>>>> phone calls to proxy.
>>>>>>>>
>>>>>>>> To your question: Sipura has a good implementation of STUN,
>>>>>>>> but has MANY options for NAT. Your problem is that the
>>> RTP
>>>>> and
>>>>>>> RTCP is not traversing
>>>>>>> the
>>>>>>>> NAT to your Sipura. Either you don't
>>> force
>>>>>>> proxying in onreply for OKs,
>>>>>>> or
>>>>>>>> something goes wrong. An ngrep trace of
>>> the
>>>>> call
>>>>>>> setup will reveal what
>>>>>>> the
>>>>>>>> problem can be.
>>>>>>>> g-)
>>>>>>>>
>>>>>>>> Sebastian Kühner wrote:
>>>>>>>>> Thank you Nils,
>>>>>>>>>
>>>>>>>>> Now it's working better!
>>>>>>>>>
>>>>>>>>> The problem that I have now is that I
>>> don't
>>>>> hear
>>>>>>> anything if I call
>>>>>>>>> from the SIPURA to a Gateway, but the
>>> callee
>>>>> is
>>>>>>> hearing me.
>>>>>>>>>
>>>>>>>>> What could be the problem of that one-way conversation? Had
>>>>>>>>> anyone of you the same problem using a Restricted Cone NAT?
>>>>>>>>>
>>>>>>>>> Thanks!
>>>>>>>>>
>>>>>>>>> Sebastian
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> ----- Original Message -----
>>>>>>>>> From: "Nils Ohlmeier"
>>> <lists at ohlmeier.org>
>>>>>>>>> To: <serusers at lists.iptel.org>
>>>>>>>>> Cc: "Sebastian Kühner"
>>> <skuehner at veraza.com>
>>>>>>>>> Sent: Tuesday, July 19, 2005 3:58 PM
>>>
>> === message truncated ===
>>
>>
>>
>>
>>
>>
>>
>>
> ___________________________________________________________________________
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>>
>
>
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