Below, I what I have gathered.  Please, note that in this case that
remote worker client, behind NAT device x-lite 4 phone,  is calling
local snom m3 phone.

Looking at the trace the snom m3 is sending rtp packets to [Remote
Worker Client Local IP] instead of sipx server.  Hence, I am not
getting sound.

I didn't include rtp packets as they are not interesting.


INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/TCP [Remote Worker Client Local
IP]:45216;branch=z9hG4bK-d8754z-52a225f7dfa4f976-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:2...@[remote Worker Client Local IP]:45216;transport=TCP>
To: <sip:[email protected]>
From: "John Smith"<sip:[email protected]>;tag=82ad6ae8
Call-ID: ZGVjOGYzMGVhYzk4YTEyYzc2ODlhMTkyMDA4ZTBiYjk.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Content-Type: application/sdp
Proxy-Authorization: Digest username=...
Supported: replaces
User-Agent: X-Lite 4 release 4.0 stamp 58832
Content-Length: 236

v=0
o=- 12936924517193114 1 IN IP4 [Remote Worker Client Local IP]
s=CounterPath X-Lite 4.0
c=IN IP4 [Remote Worker Client Local IP]
t=0 0
m=audio 49516 RTP/AVP 107 0 8 101
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

SIP/2.0 100 Trying
From: "John Smith"<sip:[email protected]>;tag=82ad6ae8
To: <sip:[email protected]>
Call-Id: ZGVjOGYzMGVhYzk4YTEyYzc2ODlhMTkyMDA4ZTBiYjk.
Cseq: 2 INVITE
Via: SIP/2.0/TCP [Remote Worker Client Local
IP]:45216;branch=z9hG4bK-d8754z-52a225f7dfa4f976-1---d8754z-;rport=63739;received=[Remote
Worker WAN IP]
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/TCP
192.168.100.122:45216;branch=z9hG4bK-d8754z-52a225f7dfa4f976-1---d8754z-;rport=63739;received=[Remote
Worker Client WAN IP]
Max-Forwards: 70
From: "Ivan Susanin" <sip:[email protected]>;tag=82ad6ae8
To: <sip:[email protected]>;tag=gsl.785ganmh
Call-Id: ZGVjOGYzMGVhYzk4YTEyYzc2ODlhMTkyMDA4ZTBiYjk.
Cseq: 2 INVITE
Contact: <sip:2...@[snom Internal Client Local IP];line=42542>
Accept: application/sdp
Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE,
NOTIFY, MESSAGE, INFO, PRACK
Content-Disposition: session
User-Agent: snom-m3-SIP/02.11 (MAC=0004132ADA0E; HW=255)
Content-Type: application/sdp
Content-Length: 311
Date: Wed, 15 Dec 2010 22:07:04 GMT

v=0
o=201 311850151 311850151 IN IP4 [SNOM Internal Client Local IP]
s=-
c=IN IP4 [SNOM Internal Client Local IP]
t=0 0
a=sendrecv
m=audio 30484 RTP/AVP 0 8 101
c=IN IP4 [SIPX Server WAN IP]
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=x-sipx-ntap:X[SIPX Server Local IP]-[SIPX Server WAN IP];186




On Mon, Dec 13, 2010 at 3:57 AM, Nikolay Kondratyev <[email protected]> wrote:
> You have to "mirror" the port on your switch.  Or use hub.
> Sip trace from snom will show sip messages only. I need to see rtp traffic
> too.
>
> Let me describe the problem I encountered some time ago and I suspect it is
> your problem now.
>
> When remote worker calls local phone, sipx relays media through itself.
> For this sake sipx sends the following sdp offer to local phone:
>
> v=0
> o=Nokia-SIPUA 63406697693656625 63406697693656625 IN IP4 192.168.12.200
> s=-
> c=IN IP4 192.168.12.200
> t=0 0
> m=audio 30098 RTP/AVP 96 0 8 97 18 98 13
> c=IN IP4 172.23.19.5
> a=sendrecv
> a=ptime:20
> a=maxptime:200
> a=fmtp:96 mode-change-neighbor=1
> a=fmtp:18 annexb=no
> a=fmtp:98 0-15
> a=rtpmap:96 AMR/8000/1
> a=rtpmap:0 PCMU/8000/1
> a=rtpmap:8 PCMA/8000/1
> a=rtpmap:97 iLBC/8000/1
> a=rtpmap:18 G729/8000/1
> a=rtpmap:98 telephone-event/8000/1
> a=rtpmap:13 CN/8000/1
> a=x-sipx-ntap:X172.23.19.5-81.211.30.104;38
>
> Note two "c" lines...
> The first "on the session level" and the second "on the media level".
> The first "c" line is remote worker address, possibly remotly nated, and not
> reachable.
> The second "c" line is ip address of sipx.
> SDP rfc says that phone MUST use the second one.
> But alas... Not all phones do that... Some still send rtp to address
> specified in the first "c" line.
>
> That's why I want to see what is happening on the wire, so that one will see
> where does snom m3 sends rtp.
>
> Rgds,
> Nikolay.
>
>> -----Original Message-----
>> From: [email protected]
>> [mailto:[email protected]] On Behalf Of
>> Roman Gelfand
>> Sent: Friday, December 10, 2010 7:03 PM
>> To: Discussion list for users of sipXecs software
>> Subject: Re: [sipx-users] Remote Workers
>>
>> I am assuming you want to see all network activity coming and
>> out of snom m3.  If so, I am not sure how to do a wireshark
>> trace on that.
>>
>> Actually, I could get a sip trace from snom m3.  Would that do?
>>
>> Thanks
>>
>> On Fri, Dec 10, 2010 at 10:36 AM, Nikolay Kondratyev
>> <[email protected]> wrote:
>> > Can you capture a trace via wireshark on the port where m3
>> is connected?
>> > The thing is that sipx does "media relay" for remote
>> workers, but sipx
>> > does it in a bit complicated way...
>> > Not all phones understand the way, sipx does "media relay".
>> > I suspect that snom does send rtp to remote worker but it
>> sends it to
>> > wrong address.
>> > Having network trace from the snom port, one will be able
>> to find out
>> > if the above is true.
>> > Rgds,
>> > Nikolay.
>> >
>> >> -----Original Message-----
>> >> From: [email protected]
>> >> [mailto:[email protected]] On Behalf Of Roman
>> >> Gelfand
>> >> Sent: Friday, December 10, 2010 5:51 PM
>> >> To: Discussion list for users of sipXecs software
>> >> Subject: Re: [sipx-users] Remote Workers
>> >>
>> >> The local phone is snom m3 and the remote phone is x-lite.
>>  I did a
>> >> tcpdump on sipx server and it shows rtp traffic proxied to
>> the snom
>> >> phone.  However, I don't see rtp traffic from snom phone to sipx.
>> >>
>> >> Please note, the external phone calls (via sip trunk) from remote
>> >> worker phone are working fine.  So, it can't be remote
>> worker phone.
>> >> It sounds more like configuration on snom is wrong.  I used sipx
>> >> server as outbound proxy on snom, but that didn't help.
>> >>
>> >> I am going to try to get you the trace logs asap.
>> >>
>> >> Thanks
>> >>
>> >> On Fri, Dec 10, 2010 at 6:27 AM, Tony Graziano
>> >> <[email protected]> wrote:
>> >> > What kind of phone? What does the registration look like?
>> >> How is the
>> >> > phone configured? Does tje remote firewall have sip alg and
>> >> spi disabled?
>> >> > ============================
>> >> > Tony Graziano, Manager
>> >> > Telephone: 434.984.8430
>> >> > Fax: 434.984.8431
>> >> >
>> >> > Email: [email protected]
>> >> >
>> >> > LAN/Telephony/Security and Control Systems Helpdesk:
>> >> > Telephone: 434.984.8426
>> >> > Fax: 434.984.8427
>> >> >
>> >> > Helpdesk Contract Customers:
>> >> > http://www.myitdepartment.net/gethelp/
>> >> >
>> >> > ----- Original Message -----
>> >> > From: [email protected]
>> >> > <[email protected]>
>> >> > To: Discussion list for users of sipXecs software
>> >> > <[email protected]>
>> >> > Sent: Thu Dec 09 11:19:24 2010
>> >> > Subject: [sipx-users] Remote Workers
>> >> >
>> >> > I register my local extension with sipx server over wan.
>> >> When I make
>> >> > an outbound (through sip trunk provider) call, everything
>> >> works great.
>> >> >  When I call local extension, they could here me but I
>> can't here
>> >> > them.  Why?
>> >> >
>> >> > Thanks in advance
>> >> >
>> >> > BTW... I have two phones registered with the same
>> extension.  Tony
>> >> > mentioned some time ago that this could cause problems.
>> >> Could this be
>> >> > one of them?
>> >> > _______________________________________________
>> >> > sipx-users mailing list
>> >> > [email protected]
>> >> > List Archive: http://list.sipfoundry.org/archive/sipx-users/
>> >> > _______________________________________________
>> >> > sipx-users mailing list
>> >> > [email protected]
>> >> > List Archive: http://list.sipfoundry.org/archive/sipx-users/
>> >> >
>> >> _______________________________________________
>> >> sipx-users mailing list
>> >> [email protected]
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>> >
>> > _______________________________________________
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>> > [email protected]
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>> >
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>
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