Roman, Why dont you send a trace? If it is too large for the list, make it available for downloading somewhere. Sipx-trace xml file filtered by call-id will not be too large. Rgds, Nikolay.
> -----Original Message----- > From: [email protected] > [mailto:[email protected]] On Behalf Of > Roman Gelfand > Sent: Thursday, December 16, 2010 10:02 PM > To: Discussion list for users of sipXecs software > Subject: Re: [sipx-users] Remote Workers > > I am a bit confused here. The two c lines are on the snom > side. The first c line is snom lan ip address. The problem > is that snom is sending rtp packets x-lite's (remote > worker's) lan ip address. The x-lite has only one c line. > > Are you saying that snom m3 should have been sending sipx's > wan ip address which is on the second c line? If so, would > this have, in turn, send rtp traffic to remote worker's wan > ip address? > > Thanks in advance > > On Thu, Dec 16, 2010 at 3:47 AM, Nikolay Kondratyev > <[email protected]> wrote: > > That is the problem is just like I described - m3 uses the first > > c-line instead of the second one. > > > > Inspite of the fact, that standard says that any sip ua > MUST use the > > second c-line, I think that the first c-line in sdp offer > is useless, > > and consequently, it is a "design mistake" in the media relay code. > > Because in practice this design (I mean two c-lines in sdp offer) > > narrows the number of compatible phones. > > But ... This is the way sipx media relay works. > > And the only way is to fix snom m3. Or change sipx media relay > > behaviour (it not in the road map, and hardly will be there). > > But snom m3 is, afaik, superceeded by m9 and snom will hardly make > > fixes for eol product. > > Alas... > > > > Rgds, > > Nikolay. > > > > > > > >> -----Original Message----- > >> From: [email protected] > >> [mailto:[email protected]] On Behalf Of Roman > >> Gelfand > >> Sent: Thursday, December 16, 2010 2:24 AM > >> To: Discussion list for users of sipXecs software > >> Subject: Re: [sipx-users] Remote Workers > >> > >> 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-;r > >> port=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] > >> >> >> 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] > >> >> 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] > >> 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] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
