thanks, I'll try it out an let you know. It matters if you have (theoretical speaking) the remote user trying to call an outside number, then the media goes like this :
remote user ---- internet ----- sipx ------ internet ------ voip provider and i want the media to go directly to provider (because of the call quality) and in case where are two remote users: remote user ------ internet ----- sipx ------- remote user On Fri, May 28, 2010 at 12:44 PM, Tony Graziano < tgrazi...@myitdepartment.net> wrote: > The only thing you can try is aggressive mode in sip to try to see if the > media will establish as peer-to-peer. > > Remote user utilizes media relay, has 2 modes (conservative & aggressive). > It requires nat be setup at your router properly for either to work. > > I don't understand why it matters, if the phone on one leg of the call is > beHind sipx, it is using the same bandwidth to the remote user whether > anchored or not. > > ============================ > Tony Graziano, Manager > Telephone: 434.984.8430 > Fax: 434.984.8431 > > Email: tgrazi...@myitdepartment.net > > 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: Irena Dolovčak <irena.dolov...@gmail.com> > To: Tony Graziano <tgrazi...@myitdepartment.net> > Cc: sipx-users@list.sipfoundry.org <sipx-users@list.sipfoundry.org> > Sent: Fri May 28 06:35:31 2010 > Subject: Re: [sipx-users] Remote office problem > > ok ok.. i think we don't understand each other completely.. > > I understand that 3cx and sipx are different. I have put 3cx just as an > example to describe to you what I want to do.. I don't even know if that > can > be done with sipx. That's what I'm trying to find out. And that's why I > have > asked you all that stuff. > > I know how to configure remote user, and I tried the aggressive mode.. but > that isn't just that what I want to accomplish.. > I don't want the sipx to act as a B2BUA.. > > how is it exactly done in sipx? can i get a step by step report what is > happening in sipx and what sipx servers and services are involved? I'm just > trying to understand what is happening and what can be done and what > can't.. > > > The only thing I am trying to accomplish is to get the phones to make calls > peer-to-peer; the sip signaling should still be servers job.. > > > > On Thu, May 27, 2010 at 11:49 AM, Tony Graziano < > tgrazi...@myitdepartment.net> wrote: > > > Consider setting up for remote users (assuming sip trunking is not > > needed). > > > > System>Internet Calling>Nat Traversal. > > > > Server>(yours)>Media Relay>advanced)>aggressive. > > > > Server>(yours)>NAT>setting the static public IP > > > > > > > > On Thu, May 27, 2010 at 5:40 AM, Tony Graziano < > > tgrazi...@myitdepartment.net> wrote: > > > >> You miss the point in that the 3cx handles nat traversal in a very > >> different way than sipx. > >> > >> With sipx, that same procedure requires use the media relay, which you > >> might adjust to be in "aggressive" mode, but there is not guarantee it > >> would > >> work in every instance, hence the default mode is "conservative". > >> > >> > >> On Thu, May 27, 2010 at 3:49 AM, Irena Dolovčak < > irena.dolov...@gmail.com > >> > wrote: > >> > >>> Hi, > >>> > >>> I don't think anchoring is the only way.. > >>> > >>> and there are more ways to bypass NAT, not just the SBC that handles > it. > >>> > >>> > >>> I have done this with 3CX PBX and mikrotik routers on both ends. > >>> > >>> I tried to reproduce the same thing in 3CX PBX, and here is the picture > >>> with steps. > >>> I'm not sure if all is 100% OK, I tried to attach the important things. > >>> > >>> I want to do the same thing in sipx to.. what are the differences > >>> between > >>> them? (the 3cx and sipx) in the structure of the pbx.. > >>> > >>> PhoneA – user > >>> PhoneB – user2 > >>> > >>> 1 –src. Address/port 192.168.0.245:5062 dst address/port > >>> 82.210.15.45:5060 > >>> phone A sends an INVITE message > >>> (sip:u...@82.210.15.45:5062 to us...@82.210.15.45) > >>> > >>> 2 – the NAT translates the src address to his public address > >>> 89.201.135.15, the src port stays the same > >>> -The router sends the packet to the internet > >>> The router gives his public ip address to eht phone. No STUN is > >>> involved. > >>> > >>> 3 – the packet comes to the second router; he sees the dst. Address > >>> 82.210.15.45 and the dst. port 5060 – he knows he must forward the > >>> packet > >>> with the dst port 5060 to the internal IP address 192.168.1.15 > >>> > >>> 4 – the 3cx PBX gets the packet on port 5060 from the ip address > >>> 89.201.135.15 > >>> It examines the packet and sees that user is trying to call user2 > >>> > >>> 5 – The PBX send the user an information packet telling him that it is > >>> trying to contact user2; > >>> The message goes back to dst address 89.201.135.15:5062 > >>> > >>> 6 – the PBX sends an INVITE message to user2 > >>> (sip:u...@82.210.15.45:5062 to us...@82.210.15.45) where the PBX tells > >>> user2 how to contact user in SDP message which is sends together with > >>> the > >>> SIP message > >>> In the SDP message, user has send information on which port should the > >>> media packages be send to. The PBX forwarded the message to user2 > >>> The message goes from src address 192.168.1.15 on port 5060 to dst > >>> address/port 192.168.1.20:5060 > >>> > >>> 7 – user2 starts to ring and sends a RINGING message back to the PBX so > >>> it knows that he is ringing; > >>> It also sends the SDP information where on which port to send the media > >>> packets > >>> Src address/port 192.168.1.20:5060 dst address/port 192.168.1.15:5060 > >>> > >>> 8 – the PBX gets the message from user2 and forwards it back to user so > >>> he allso knows where to send media packets; it also sends the ring > alert > >>> to > >>> user so the user knows user2 is ringing > >>> The packet is send to the router with Src address/port > >>> 192.168.1.15:5060dst address/port > >>> 89.201.135.15:5062 > >>> > >>> 9 – the router gets the packet form the PBX and sees the dst > >>> address/port > >>> 89.201.135.15:5062; to be able to send the packet, he traverses the > src > >>> address 192.168.1.15 to 82.210.15.45 and sends it to the internet > >>> > >>> 10 – the second router gets the message now with src address > >>> 82.210.15.45:5060 and dst address 89.201.135.15:5062; he knows he has > to > >>> forward the message back to private ip address 192.168.0.245:5062 > >>> > >>> 11 – when the user2 answers the call, the phone sends an OK message > that > >>> it has picked up the phone. (The process is the same as at step 10) > >>> > >>> 12 – the phoneB sends the RTP packet direckt to the phoneA as it has > all > >>> necessary information as the Ip address and port to which he should > >>> send > >>> it. > >>> Src address: 192.168.1.20:9000 dst address 89.201.135.15:11000 > >>> the router gets the packet, translates the private IP to 82.210.15.45 > >>> and send it to 89.201.135.15:11000 > >>> The second router gets the packet with src address/port > >>> 82.210.15.45:9000 and dst address/port 89.201.135.15:11000 and it > knows > >>> that it must forward it to the private ip address 192.168.0.245:11000 > >>> > >>> 13 – when user hang up, the phone sends a BYE message to the PBX to > >>> alert > >>> user2 > >>> Src address/port 192.168.0.245:5062 dst address/port 82.210.15.45:5060 > >>> > >>> 14 – the router gets the message, it outs a new src address > >>> (89.201.135.15) and send it to the dst address 82.210.15.45 > >>> > >>> 15 – the router gets the packet with dst address 82.210.15.45:5060 and > >>> it forwards it to internal ip 192.168.1.15 becouse it is tols to send > >>> the > >>> message with dst port 5060 to that ip address > >>> > >>> 16 – the PBX gets the message, and sends the message to user2 on ip > >>> address 192.168.1.20 > >>> > >>> 17- user2 sends an ACK message to the PBX. The call ends > >>> > >>> > >>> > >>> > >>> On Fri, May 21, 2010 at 1:55 PM, Tony Graziano < > >>> tgrazi...@myitdepartment.net> wrote: > >>> > >>>> Yes, vpn the remote site and it will take the media peer-to-peer. > >>>> > >>>> You seem to be stuck on the fact the sipx must anchor the media and > fpr > >>>> whatever reason don't want to do that. It ONLY ANCHORS the media for > >>>> the > >>>> calls that go through NAT, it won't anchor the media for you > lan-to-lan > >>>> calls. > >>>> > >>>> In SIP, anytime you cross NAT something needs to anchor the media. > That > >>>> device is called a SBC. > >>>> > >>>> I think if you want it to work some other way using a SIP (not sipx) > >>>> based > >>>> system, you need to invent your own technology. > >>>> > >>>> Good luck. > >>>> ============================ > >>>> Tony Graziano, Manager > >>>> Telephone: 434.984.8430 > >>>> Fax: 434.984.8431 > >>>> > >>>> Email: tgrazi...@myitdepartment.net > >>>> > >>>> 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: Irena Dolovčak <irena.dolov...@gmail.com> > >>>> To: Tony Graziano <tgrazi...@myitdepartment.net> > >>>> Sent: Fri May 21 07:40:57 2010 > >>>> Subject: Re: [sipx-users] Remote office problem > >>>> > >>>> I'm not sure am I following you. > >>>> > >>>> Are you saying that the sipx can only be configured to anchor the > media > >>>> (as > >>>> a B2BUA)? Is there no other solution? > >>>> I understand that the sipxbridge anchor the media, that's why I > removed > >>>> it. > >>>> > >>>> Is there any solution that I could use to make the remote worker to > >>>> connect > >>>> in a by-pass mode and not trunked? > >>>> > >>>> > >>>> > > >>>> > I thbik it you have remote workers enabled and server behind nat > >>>> > configured > >>>> > "it should work". Right now sipx is being told to route to ports it > >>>> > is > >>>> not > >>>> > employing because the sipxbridge component is not being used not is > >>>> the > >>>> > remote workers aspect. If you support either, you define an SBC, you > >>>> have > >>>> > none. That's what sipxbridge does. Ingate makes a nice one too. > >>>> > > >>>> > > >>>> What exactly do you mean by "it should work"? > >>>> > >>>> -- > >>>> Irena Dolovčak > >>>> > >>> > >>> > >>> > >>> -- > >>> Irena Dolovčak > >>> > >> > >> > >> > >> -- > >> ====================== > >> Tony Graziano, Manager > >> Telephone: 434.984.8430 > >> sip: tgrazi...@voice.myitdepartment.net > >> Fax: 434.984.8431 > >> > >> Email: tgrazi...@myitdepartment.net > >> > >> LAN/Telephony/Security and Control Systems Helpdesk: > >> Telephone: 434.984.8426 > >> sip: helpd...@voice.myitdepartment.net > >> Fax: 434.984.8427 > >> > >> Helpdesk Contract Customers: > >> http://www.myitdepartment.net/gethelp/ > >> > >> Why do mathematicians always confuse Halloween and Christmas? > >> Because 31 Oct = 25 Dec. > >> > >> > > > > > > -- > > ====================== > > Tony Graziano, Manager > > Telephone: 434.984.8430 > > sip: tgrazi...@voice.myitdepartment.net > > Fax: 434.984.8431 > > > > Email: tgrazi...@myitdepartment.net > > > > LAN/Telephony/Security and Control Systems Helpdesk: > > Telephone: 434.984.8426 > > sip: helpd...@voice.myitdepartment.net > > Fax: 434.984.8427 > > > > Helpdesk Contract Customers: > > http://www.myitdepartment.net/gethelp/ > > > > Why do mathematicians always confuse Halloween and Christmas? > > Because 31 Oct = 25 Dec. > > > > > > > -- > Irena Dolovčak > -- Irena Dolovčak
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