Hello Yasuro,
what I am wondering about is: Freeswitch is offering port 16670 in its
SDP message of the SIP OK message, and 192.168.11.250 is trying to send
RTP packets to 16671 instead. But this Freeswitch port is not open. So
192.168.11.250 closes the connection: See
217 49.351287 192.168.11.250 5005 192.168.11.11
16671 RTCP Receiver Report Source description
218 49.351369 192.168.11.11 5005 192.168.11.250
16671 ICMP Destination unreachable (Port unreachable)
--
With kind regards
Peter Steinbach
Telefaks Services GmbH
Theo-Geisel-Strasse 25
D 61250 Usingen, Germany
mailto:lists (att)
telefaks.deInternet:
www.telefaks.deYasuro schrieb:
Michael and other FreeSWITCH gurus:
As per your instruction, I have created logs. Please take a look
<http://pastebin.freeswitch.org/14479>. It includes FS's messages
while it initializes itself. The VoIP adapter (SIP gateway, which also
works as a router)'s IP address is *192.168.11.250* on the LAN side.
That of the PC which is running FS is *192.168.11.11*. Its firewall
function was turned off during this testing. The summary of packet
exchange between the two during this same period is here
<http://pastebin.freeswitch.org/14480>. Please see my original posting
(which is included at the end of this message) for the setup of my
home LAN.
~250 is sending RTCP receiver report before ~11 starts RTP
communication. I am wondering if the former is giving up on the call
because of time out; and, if it is true, if there is any way to have
it wait longer.
I am completely stuck about this issue and I'll welcome any input you
have.
Thanks!
Yasuro
Michael Collins wrote (11/12/2010 4:43 AM):
Run your same test on FreeSWITCH but turn on sip debugging at the fs_cli:
sofia global siptrace on
Then make the call and capture the output and put into new pastebin.
Hopefully you'll see why the channel is already hungup when it goes
to play music.
-MC
2010/11/10 Yasuro <yasuro@yasuro.com <mailto:yasuro@yasuro.com>>
Hi, FreeSWITCH gurus! I need your help!
First off, I am new to FS and I am new to Internet telephony as
well. Heck, I am new to the concept of NAT, UPnP, etc., so please
bear with my ignorance.
I subscribe to a VoIP service at home, with which I get one DID.
They supply me a VoIP adapter. Their expected usage is for you to
plug in analog phones to the analog phone jacks in the VoIP
adapter. However, It also has four Ethernet LAN ports and it acts
as a router. You can also access it from the LAN side and
register with its built-in SIP gateway.
What I would like to do is to run FS (Windows version) on one of
the Windows PCs, have it register with the SIP gateway, and have
it act as an AA or IVR. For testing, I am having it just play music.
When I tried the same idea with AsterikWin32, it worked just as I
had hoped; it answered incoming calls automatically. However, I
somehow cannot make it work with FS. I simulate incoming calls to
my DID number with Skype's Sypeout. It fails after a short while
with such error messages as "network error." It appears the call
was never answered.
FS is assigned an extension number 7 at the gateway. When I call
extension 7 from a different extension (at the gateway level, not
an extension inside FS), FS does answer the call and I hear
music. FS fails to answer only incoming calls from outside.
I think my FS configuration is fairly standard. I created an
external SIP profile for the gateway under
conf/sip_profiles/external/ and modified
conf/dialplan/public/00_inbound_did.xml so incoming calls to the
gateway will be transferred to an extension within FS.
FS's messages and logs, plus the result of packet captures
indicate that FS /thinks /it has answered the call, and goes on
to initiate media communication. I see RTP packets going from FS
to the SIP gateway. What's different from AsteriskWin32's case is
that there are no RTP packets coming back from the SIP gateway to
FS. Turning of the firewall of the PC does not seem to change the
result in any way.
For your perusal, I have created the following logs of
communication between FS/AsteriskWin32 and the SIP gateway:
* AsteriskWin32's case
o Summary: http://pastebin.freeswitch.org/14457
o Details: http://pastebin.freeswitch.org/14460
* FreeSWITCh's case
o Summary: http://pastebin.freeswitch.org/14462
o Details: http://pastebin.freeswitch.org/14463
o Log: http://pastebin.freeswitch.org/14465
The IP address of the SIP gateway is *192.168.11.250*, and that
of the PC FS/AsteriskWin32 resides in is *192.168.11.11*. My DID
number is masked as ABCDEFGHIJ. I do not know if it gives you any
useful information, but those files include the registration
phase. /FS's log was taken at a different time/, so it does not
entirely match the packet captures.
I also have the corresponding Pcap files. Please let me know if
you need them.
I am not entirely sure, but I think as far as what I'd like to do
is concerned, NAT is not going to be an issue, because the
FS/AsteriskWin32 PC and the SIP gateway (its LAN side IP address)
are on the same subnet (192.168.11/24). At this time, I do not
need to access FS from the Internet.
Finally, I will give you more details about my setup, which may
or may not be relevant to this issue.
My home LAN is set up this way:
http://i139.photobucket.com/albums/q302/starplatina/For%20Blogs/HomeLANSetup.jpg
Please note that there are /two layers/ of NAT, and that in the
inner layer, two NAT devices exist. I know it looks convoluted,
but there are logical reasons for this setup.
The VoIP service provider only supports the PCMU codec. The music
file I prepared for this testing is encoded in PCMU, so codecs
will not be an issue.
Please do not hesitate to ask if you have any questions. Thanks
for your help in advance!
Yasuro
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