21 Sep
2007
21 Sep
'07
2:24 p.m.
The error was in Mediaproxy.
In rtphandler.py i changed
nonPublicNetworks = [
{'name': '0.0.0.0', 'value': 0x00000000L, 'mask':
0xff000000L},
{'name': '10.0.0.0', 'value': 0x0a000000L, 'mask':
0xff000000L},
{'name': '127.0.0.0', 'value': 0x7f000000L, 'mask':
0xff000000L},
{'name': '172.16.0.0', 'value': 0xac100000L, 'mask':
0xfff00000L},
{'name': '192.168.0.0', 'value': 0xc0a80000L, 'mask':
0xffff0000L},
{'name': '224.0.0.0', 'value': 0xe0000000L, 'mask':
0xf0000000L}
]
To
nonPublicNetworks = [
{'name': '0.0.0.0', 'value': 0x00000000L, 'mask':
0xff000000L},
{'name': '10.0.0.0', 'value': 0x0a000000L, 'mask':
0xff000000L},
{'name': '127.0.0.0', 'value': 0x7f000000L, 'mask':
0xff000000L},
{'name': '224.0.0.0', 'value': 0xe0000000L, 'mask':
0xf0000000L}
]
I think Mediaproxy got confused with the RFC1918 IP's. In my setup
there is no NAT between 172.17.0.0/16 and 192.168.0.0/24 - just a
router.
On 9/21/07, Norman Brandinger <norm(a)goes.com> wrote:
Is there a firewall in the picture ? You have two
different subnets and
there probably is a box doing some (NAT) translation / routing between
them. Is is possible the RTP stream is being blocked at the firewall ?
Norm
Morten Isaksen wrote:
--
Morten Isaksen
http://www.misak.dk/blog/
Hi!
I can see in the mediaproxy log the it is initialized to proxy the
call, but I newer get a "session xxxxx: called signed in from xxx"
from Asterisk.
session.py shows that the the connection between mediaproxy and
Asterisk is missing.
I will try to take a look at the sip debug from asterisk and try to
change the NAT settings in Asterisk.
Thanks for your input.
On 9/20/07, Norman Brandinger <norm(a)goes.com> wrote:
You stated that you've forced every call
through mediaproxy. Are you
positive ?
Have you taken a look at the mediaproxy logs (and/or sessions.py when
the call is up) ? They might provide some useful information to you.
Ditto for Asterisk "sip set debug on" (note that the sip debug command
format is a moving target).
Have you looked at the "nat=" settings in sip.conf as well ? At times,
they tie closely with "canreinvite=".
Norm
Morten Isaksen wrote:
Hi!
canreinvite is set to no and the OpenSER/mediaproxy is "hiding" the
clients IP-addresses from Asterisk, so I am pretty sure that this is
not the issue.
On 9/20/07, Norman Brandinger <norm(a)goes.com> wrote:
> Hi Morten,
>
> Admittedly, I haven't looked closely at your trace. However, based on
> the description you gave, the first place to look is at the "canrevite"
> setting in Asterisk sip.conf. You might want to try "canreinvite=no"
> after reading up on this particular setting.
>
> Regards,
> Norm
>
>
> Morten Isaksen wrote:
>
>
>> Hi!
>>
>> I have a strange problem with a missing RTP stream between OpenSER and
>> Asterisk. I am not sure if it is OpenSER og Asterisk related.
>>
>> I have this setup
>>
>> Phone A (172.17.96.17) --
>> \ Openser -- Asterisk
>> -- PSTN
>> / (192.168.0.6) (192.168.0.3)
>> Phone B (172.17.96.10) -- (172.17.64.1)
>>
>> I also have a Mediaproxy running on OpenSER and I force every call to
>> use the Mediaproxy.
>>
>> I call from Phone A or B to the PSTN works fine and from PSTN to Phone
>> A or B it also works.
>>
>> I have the dialplan logic on my Asterisk server so I want calls from
>> Phone A to Phone B to pass the Asterisk server. And this is were I
>> have the problem. When the call is established the RTP stream is
>> missing between Mediaproxy and Asterisk. I only have a RTP stream
>> between the phones and Mediaproxy. As far as I can see the SIP
>> signalling is correct.
>>
>> The SIP traces is listed below. Can you spot the problem in this?
>>
>> I will buy a beer (or 5) at OpenSER training in Rome to anyone who can
>> help me solve this problem.
>>
>> SIP trace between the phones and OpenSER:
>>
>>
>>
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Morten Isaksen