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/
