[Freeswitch-users] Help! FS Unable to Handle Incoming Calls Through SIP Gateway
David Ponzone
david.ponzone at ipeva.fr
Sun Nov 14 15:34:45 PST 2010
Peter,
that's not RTP, but RTCP.
The RTCP port is always RTP+1.
David Ponzone Direction Technique
email: david.ponzone at ipeva.fr
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Le 14/11/2010 à 22:31, Peter Steinbach a écrit :
> 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.de
> Internet: www.telefaks.de
>
>
>
> Yasuro 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 at yasuro.com <mailto:yasuro at 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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>>
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