Re: [asterisk-users] How to use stun server?
I'm sure there was a perfectly good reason for encoding the devices IP address inside the SIP data when they invented it, but right now, I can't think why one thing i still dont understnd. if the device we are using is a computer, and we r running a softphone on it. and side by side we are also surfing the net. then why is it so that web content is coming into the computer without any problem but rtp data is not. i think both the web application and softphone are using computer's local ip address in their requests. So whats the reason for this? I understand how stun works but thanx for explaining it in so simple and concise way. One other question which has been bothering me is: If the client phone is behind nat, that means there is NATTING going on between public internet and local net. Then why do we need stun? NATTING should handle the problem itself as it does for other applications running on the same computer where softphone is also running. On 8/2/07, Gordon Henderson [EMAIL PROTECTED] wrote: On Thu, 2 Aug 2007, Rizwan Hisham wrote: hi again.well i have been trying to know what is the relationship between asterisk and stun. what i mean is, i understand that a client requests stun server to know whether its behind a nat or not. if its not, then its ok. if it is behind nat, then what? Now client knows what kind of nat it is behind, what is the roll of asterisk in it. asterisk already knows client's public ip whether its behind nat or not, if the client is registered. So how does stun simplify things if there are nat problems. There is no relationship between asterisk and STUN. After requesting stun server and recieving the required information from stun server.what happens next? I hope im clear in stating my problem. I'm not a STUN/SIP protocol gury by any means, but this is my understanding (and it might be a bit simplistic) When something communicates with something else using SIP, the sending device (eg phone) puts it's own IP address inside the SIP data packet. That IP address is the IP address of the device - it doesn't know anything about anything else, just the IP address it has. This would work well if NAT hadn't been invented, unfortunately it was. The listening side (eg. asterisk), extracts this IP address and uses it to send data back. So if the originating device is behind NAT, and it's on (eg) 192.168.0.42 then the other end, gets that IP address and tries to send data back to it. Which, as 192.168.0.42 is on a private network, it can't do. Oops. So the original device uses a STUN server to poke a few bytes over the interweb and the STUN server replys back with some information - such as the real external IP address and port numbers it's using. The STUN server is a tiny bit of software running on a host somewhere with a real IP address (or 2!) and is (or can be) quite independant of the asterisk server. Original device can then put those values returned from the STUN server inside the SIP data packets (rather than it's 'real' natted IP address) and send them off to the other end, which can then use them to send the replys back to. The device should only need to access the STUN server once in it's life, but devices periodically check, just in-case things have changed. They do not relay data through the STUN server. So that's for device to asterisk box. Asterisk boxes are supposed to be directly connected to the Internet with no NAT and a real live IP address. (or at least that's the best possible way to do it!) If they aren't ... Then the first thing you need to do is arrange port-forwarding from the firewall to the asterisk box. You'll need to forward the ports you need - eg. for SIP it might be 5060-5069 and for RTP it might be 1-2. But the asterisk server still needs to know what it's real external IP address is so it can put that in the SIP packets rather than it's own NATted address, and as asterisk can't use a STUN server, you need to explicitly tell it - this is in the sip.conf file and looks like: nat=yes localnet=192.168.2.0/24 externip=1.2.3.4 So now the asterisk server knows that anything that originates from the local network doesn't need to be translated, but anything going out needs to have the SIP data re-written with the real external IP address. Now (AIUI) some SIP proxys can look inside SIP data packets and see that the IP address given by the device is not the same as the IP address that the packet came from and adjust things accordingly.. Asterisk, not being a SIP proxy doesn't do this, so if your phone is talking to a server via a proxy, then you may not need to tell the phone about a STUN server. The people running the asterisk+SIP proxy will tell you if this is the case. I'm sure there was a perfectly good reason for encoding the devices IP address inside the SIP data when they invented it, but right now, I can't think why... See
Re: [asterisk-users] How to use stun server?
El Fri, Aug 03 de 2007 a las 20:24 +0500, Rizwan Hisham comentaba: I'm sure there was a perfectly good reason for encoding the devices IP address inside the SIP data when they invented it, but right now, I can't think why one thing i still dont understnd. if the device we are using is a computer, and we r running a softphone on it. and side by side we are also surfing the net. then why is it so that web content is coming into the computer without any problem but rtp data is not. i think both the web application and softphone are using computer's local ip address in their requests. So whats the reason for this? Simple, web content uses TCP while RTP uses UDP to carry the data. In TCP your computer needs to establish a connection with the remote side before each one can send any data, the device which is doing the NAT realizes that and creates a bridge between your computer's IP/Port and the remote site IP/Port. In the case of SIP/RTP over UDP is different. Your softphone sends the signaling over a UDP port, the remote site receives the data and responses back to the IP/Port it recived it from (the IP/Port of the NATing device), the device which is doing the NAT knows that you have recently send data over that IP/Port and routes it back to you. Thats why SIP signaling can work fine even behind a NAT (nat=yes). RTP flow is also different. Your softphone specify it wants to receive RTP in a IP/Port (private IP/Port), when the remote site wants to send you RTP data it cannot be routed because that address is private, it cannot send the data to the address of the NATing device because the port this device is using for your outgoing RTP is different than the port you specified. So the RTP that is destinated to you gets lost. I understand how stun works but thanx for explaining it in so simple and concise way. One other question which has been bothering me is: If the client phone is behind nat, that means there is NATTING going on between public internet and local net. Then why do we need stun? NATTING should handle the problem itself as it does for other applications running on the same computer where softphone is also running. NATting can, in someway, handle the problem when you originate the call, but it cannot do it when someone wants to reach you later. The SIP header Contact is used for this, when someone wants to reach you it uses the address you specified in that header, so it must be a public IP address which you obtained from the STUN server or another mean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use stun server?
Well, there is a valid reason for embedding the IP and port inside SIP. First, SIP can be proxied multiple times (a chain, the proxy of your provider talks to other proxy and so on until you reach the other person provider's proxy), this is the way it was designed and still used, so if you detect the IP address using layer 3 means, you may be connecting to a proxy, no the endpoint. The SIP conversation is always relayed via the proxy chain (never directly), and the proxies add/delete things as the proxy passes thru them. The ip and port embedded inside SIP tell the endpoints where is the real other endpoint. Second, SIP as it's name indicates Session Iniciation Protocol, is not a media protocol, in that case it would be called something like voice carry protocol; SIP only carries the signaling to create/destroy media sessions, capabilities negociation, etc (not only voice, but multple video, games, messaging, presence,...), but doesn't carry the media. Also SIP normally is implemented in TCP and this is planly BAD for media streams, because of the delays and retries that tcp incurrs to deliver a ordered and complete stream of data. So the media streams can't be carried over tcp because of it's time-delay sensibility. So? you carry media sessions over UDP. So, if a packet doesn't arrive? sorry it's lost forever. a packet arrives late? simple, drop it, but doesn't stops the delivery of the future packets, as if you did it over tcp. The RTP packets go end to end directly, not via the proxy chain, using a shorter route, and usually carrying TOS flags and are applied QoS. Adobe flash video streaming is a example of doing strange things; to be compatible with web an browsers, they did an implementation of a RTP protocol over TCP (the server has to explicitly watch receipts timestamps, and only transmit the most up to date data, skiping delayed data), usually works fine, but when things go wrong, they go VERY wrong and the video suffers until you can clean the bottleneck. Also there is no obligation of using RTP as a real time media companion to SIP, you could use SRTP, ZRTP, or other propietary prococol; for other things like games, the only games in town are propietary media protocols. This works wonderfully in the plain old internet, but in the NAT'ed paranoid internet of our days, the clients put their private address in the SIP fields, and the other end can only laugh because the address is unreachable. There is the usefulness of STUN and TURN, to detect you NAT type and external ip address and port so the incoming RTP stream can reach the endpoint via a NAT provided hole (ip address and port that forwards to the endpoint [internal] ip address and port). Also an administrator can put a proxy in the network's boundary that rewrites SIP requests and eliminate the use of STUN. But problems don't end there, for certain types of NAT, firewalls and network setups, there is NO incomming way. This is the point where things become really difficult. That is the use of STUN. You should read a little more to understand the basics of the prococols, more if you work for an ITSP. Message: 1 Date: Fri, 3 Aug 2007 20:24:17 +0500 From: Rizwan Hisham [EMAIL PROTECTED] Subject: Re: [asterisk-users] How to use stun server? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 I'm sure there was a perfectly good reason for encoding the devices IP address inside the SIP data when they invented it, but right now, I can't think why one thing i still dont understnd. if the device we are using is a computer, and we r running a softphone on it. and side by side we are also surfing the net. then why is it so that web content is coming into the computer without any problem but rtp data is not. i think both the web application and softphone are using computer's local ip address in their requests. So whats the reason for this? I understand how stun works but thanx for explaining it in so simple and concise way. One other question which has been bothering me is: If the client phone is behind nat, that means there is NATTING going on between public internet and local net. Then why do we need stun? NATTING should handle the problem itself as it does for other applications running on the same computer where softphone is also running. On 8/2/07, Gordon Henderson [EMAIL PROTECTED] wrote: On Thu, 2 Aug 2007, Rizwan Hisham wrote: hi again.well i have been trying to know what is the relationship between asterisk and stun. what i mean is, i understand that a client requests stun server to know whether its behind a nat or not. if its not, then its ok. if it is behind nat, then what? Now client knows what kind of nat it is behind, what is the roll of asterisk in it. asterisk already knows client's public ip whether its behind nat or not, if the client
Re: [asterisk-users] How to use stun server?
hi again.well i have been trying to know what is the relationship between asterisk and stun. what i mean is, i understand that a client requests stun server to know whether its behind a nat or not. if its not, then its ok. if it is behind nat, then what? Now client knows what kind of nat it is behind, what is the roll of asterisk in it. asterisk already knows client's public ip whether its behind nat or not, if the client is registered. So how does stun simplify things if there are nat problems. After requesting stun server and recieving the required information from stun server.what happens next? I hope im clear in stating my problem. Hope to hear from you soon On 8/1/07, SIP [EMAIL PROTECTED] wrote: No... there's no STUN server built into Asterisk. Asterisk handles NAT in a different way... and is an endpoint rather than a proxy, so it doesn't really NEED STUN built into it. However, we run a STUN server on the same machine as an Asterisk server and see nothing in terms of load increase. STUN's footprint is rather negligible. N. Rizwan Hisham wrote: Ok thanx. One more thing to ask is: does asterisk has a stun server implemented in it or not. i mean does asterisk contain a stun server and provides any application which can be used for enabling the stun server in asterisk? On 8/1/07, *SIP* [EMAIL PROTECTED] mailto: [EMAIL PROTECTED] wrote: STUN is a pretty simplistic server. There's nothing else that needs to be configured on the STUN server side. It's pretty much either running or it's not. Just start plugging in the server to your clients and give it a whirl. It should work. N. Rizwan Hisham wrote: Hi all, This is the first time i am using stun with asterisk for nat problems. I have read the rfc which describes how stun works. i didnt have any problems understanding it. I have also intalled the stun server called stund which i downloaded from sourceforge. I have seen on the list that most people use stund here. I have started the stun server and its running silently. Now i dont know what to do next. Is there something more to it, like configuration files which i can use for special configuration for asterisk, or is there not. How do i proceed, if there is nothing more to configure in stun, does that mean i can start configuring my clinets (xten and sipura) to use stun server? -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com http://www.axvoice.com http://www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com http://www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use stun server?
Honestly, it's really up to the client how it handles information from STUN. Ideally, what will happen is that it will modify its Contact headers and SDP information to include the STUN-discovered IP address and port. In so doing, when it sends out a request to another server, that server will then know the proper IP address to use to send data back to the UA. This is primarily of importance when you are using SER/OpenSER as a SIP proxy, or have Asterisk set to canreinvite=yes What happens is that this allows clients to directly talk to each other using publicly-addressable IP addresses, taking Asterisk out of the equation except for passing signaling information. It can save bandwidth. It can ease Asterisk load. Etc, etc. If you have canreinvite=no set on your Asterisk server, and you're using Asterisk for your SIP communications, then STUN will still inform the UA to rewrite its appropriate headers, but you'll see no real difference. Audio will still be bridged by the Asterisk box. Your bandwidth won't change. Etc, etc. It all really depends on what you want to get out of this whole thing and what your overall network design is. N. Rizwan Hisham wrote: hi again.well i have been trying to know what is the relationship between asterisk and stun. what i mean is, i understand that a client requests stun server to know whether its behind a nat or not. if its not, then its ok. if it is behind nat, then what? Now client knows what kind of nat it is behind, what is the roll of asterisk in it. asterisk already knows client's public ip whether its behind nat or not, if the client is registered. So how does stun simplify things if there are nat problems. After requesting stun server and recieving the required information from stun server.what happens next? I hope im clear in stating my problem. Hope to hear from you soon On 8/1/07, *SIP* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: No... there's no STUN server built into Asterisk. Asterisk handles NAT in a different way... and is an endpoint rather than a proxy, so it doesn't really NEED STUN built into it. However, we run a STUN server on the same machine as an Asterisk server and see nothing in terms of load increase. STUN's footprint is rather negligible. N. Rizwan Hisham wrote: Ok thanx. One more thing to ask is: does asterisk has a stun server implemented in it or not. i mean does asterisk contain a stun server and provides any application which can be used for enabling the stun server in asterisk? On 8/1/07, *SIP* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: STUN is a pretty simplistic server. There's nothing else that needs to be configured on the STUN server side. It's pretty much either running or it's not. Just start plugging in the server to your clients and give it a whirl. It should work. N. Rizwan Hisham wrote: Hi all, This is the first time i am using stun with asterisk for nat problems. I have read the rfc which describes how stun works. i didnt have any problems understanding it. I have also intalled the stun server called stund which i downloaded from sourceforge. I have seen on the list that most people use stund here. I have started the stun server and its running silently. Now i dont know what to do next. Is there something more to it, like configuration files which i can use for special configuration for asterisk, or is there not. How do i proceed, if there is nothing more to configure in stun, does that mean i can start configuring my clinets (xten and sipura) to use stun server? -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com http://www.axvoice.com http://www.axvoice.com http://www.axvoice.com http://www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] How to use stun server?
On Thu, 2 Aug 2007, Rizwan Hisham wrote: hi again.well i have been trying to know what is the relationship between asterisk and stun. what i mean is, i understand that a client requests stun server to know whether its behind a nat or not. if its not, then its ok. if it is behind nat, then what? Now client knows what kind of nat it is behind, what is the roll of asterisk in it. asterisk already knows client's public ip whether its behind nat or not, if the client is registered. So how does stun simplify things if there are nat problems. There is no relationship between asterisk and STUN. After requesting stun server and recieving the required information from stun server.what happens next? I hope im clear in stating my problem. I'm not a STUN/SIP protocol gury by any means, but this is my understanding (and it might be a bit simplistic) When something communicates with something else using SIP, the sending device (eg phone) puts it's own IP address inside the SIP data packet. That IP address is the IP address of the device - it doesn't know anything about anything else, just the IP address it has. This would work well if NAT hadn't been invented, unfortunately it was. The listening side (eg. asterisk), extracts this IP address and uses it to send data back. So if the originating device is behind NAT, and it's on (eg) 192.168.0.42 then the other end, gets that IP address and tries to send data back to it. Which, as 192.168.0.42 is on a private network, it can't do. Oops. So the original device uses a STUN server to poke a few bytes over the interweb and the STUN server replys back with some information - such as the real external IP address and port numbers it's using. The STUN server is a tiny bit of software running on a host somewhere with a real IP address (or 2!) and is (or can be) quite independant of the asterisk server. Original device can then put those values returned from the STUN server inside the SIP data packets (rather than it's 'real' natted IP address) and send them off to the other end, which can then use them to send the replys back to. The device should only need to access the STUN server once in it's life, but devices periodically check, just in-case things have changed. They do not relay data through the STUN server. So that's for device to asterisk box. Asterisk boxes are supposed to be directly connected to the Internet with no NAT and a real live IP address. (or at least that's the best possible way to do it!) If they aren't ... Then the first thing you need to do is arrange port-forwarding from the firewall to the asterisk box. You'll need to forward the ports you need - eg. for SIP it might be 5060-5069 and for RTP it might be 1-2. But the asterisk server still needs to know what it's real external IP address is so it can put that in the SIP packets rather than it's own NATted address, and as asterisk can't use a STUN server, you need to explicitly tell it - this is in the sip.conf file and looks like: nat=yes localnet=192.168.2.0/24 externip=1.2.3.4 So now the asterisk server knows that anything that originates from the local network doesn't need to be translated, but anything going out needs to have the SIP data re-written with the real external IP address. Now (AIUI) some SIP proxys can look inside SIP data packets and see that the IP address given by the device is not the same as the IP address that the packet came from and adjust things accordingly.. Asterisk, not being a SIP proxy doesn't do this, so if your phone is talking to a server via a proxy, then you may not need to tell the phone about a STUN server. The people running the asterisk+SIP proxy will tell you if this is the case. I'm sure there was a perfectly good reason for encoding the devices IP address inside the SIP data when they invented it, but right now, I can't think why... See http://www.ietf.org/rfc/rfc3261.txt for the details! Gordon ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to use stun server?
Hi all, This is the first time i am using stun with asterisk for nat problems. I have read the rfc which describes how stun works. i didnt have any problems understanding it. I have also intalled the stun server called stund which i downloaded from sourceforge. I have seen on the list that most people use stund here. I have started the stun server and its running silently. Now i dont know what to do next. Is there something more to it, like configuration files which i can use for special configuration for asterisk, or is there not. How do i proceed, if there is nothing more to configure in stun, does that mean i can start configuring my clinets (xten and sipura) to use stun server? -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use stun server?
STUN is a pretty simplistic server. There's nothing else that needs to be configured on the STUN server side. It's pretty much either running or it's not. Just start plugging in the server to your clients and give it a whirl. It should work. N. Rizwan Hisham wrote: Hi all, This is the first time i am using stun with asterisk for nat problems. I have read the rfc which describes how stun works. i didnt have any problems understanding it. I have also intalled the stun server called stund which i downloaded from sourceforge. I have seen on the list that most people use stund here. I have started the stun server and its running silently. Now i dont know what to do next. Is there something more to it, like configuration files which i can use for special configuration for asterisk, or is there not. How do i proceed, if there is nothing more to configure in stun, does that mean i can start configuring my clinets (xten and sipura) to use stun server? -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com http://www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use stun server?
I cannot really help except to say you may want to ask this question on the stund list (if they have one) since it relates more to the STUN software than it does Asterisk. Thanks, Steve Rizwan Hisham wrote: Hi all, This is the first time i am using stun with asterisk for nat problems. I have read the rfc which describes how stun works. i didnt have any problems understanding it. I have also intalled the stun server called stund which i downloaded from sourceforge. I have seen on the list that most people use stund here. I have started the stun server and its running silently. Now i dont know what to do next. Is there something more to it, like configuration files which i can use for special configuration for asterisk, or is there not. How do i proceed, if there is nothing more to configure in stun, does that mean i can start configuring my clinets (xten and sipura) to use stun server? -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com http://www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use stun server?
Ok thanx. One more thing to ask is: does asterisk has a stun server implemented in it or not. i mean does asterisk contain a stun server and provides any application which can be used for enabling the stun server in asterisk? On 8/1/07, SIP [EMAIL PROTECTED] wrote: STUN is a pretty simplistic server. There's nothing else that needs to be configured on the STUN server side. It's pretty much either running or it's not. Just start plugging in the server to your clients and give it a whirl. It should work. N. Rizwan Hisham wrote: Hi all, This is the first time i am using stun with asterisk for nat problems. I have read the rfc which describes how stun works. i didnt have any problems understanding it. I have also intalled the stun server called stund which i downloaded from sourceforge. I have seen on the list that most people use stund here. I have started the stun server and its running silently. Now i dont know what to do next. Is there something more to it, like configuration files which i can use for special configuration for asterisk, or is there not. How do i proceed, if there is nothing more to configure in stun, does that mean i can start configuring my clinets (xten and sipura) to use stun server? -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com http://www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use stun server?
No... there's no STUN server built into Asterisk. Asterisk handles NAT in a different way... and is an endpoint rather than a proxy, so it doesn't really NEED STUN built into it. However, we run a STUN server on the same machine as an Asterisk server and see nothing in terms of load increase. STUN's footprint is rather negligible. N. Rizwan Hisham wrote: Ok thanx. One more thing to ask is: does asterisk has a stun server implemented in it or not. i mean does asterisk contain a stun server and provides any application which can be used for enabling the stun server in asterisk? On 8/1/07, *SIP* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: STUN is a pretty simplistic server. There's nothing else that needs to be configured on the STUN server side. It's pretty much either running or it's not. Just start plugging in the server to your clients and give it a whirl. It should work. N. Rizwan Hisham wrote: Hi all, This is the first time i am using stun with asterisk for nat problems. I have read the rfc which describes how stun works. i didnt have any problems understanding it. I have also intalled the stun server called stund which i downloaded from sourceforge. I have seen on the list that most people use stund here. I have started the stun server and its running silently. Now i dont know what to do next. Is there something more to it, like configuration files which i can use for special configuration for asterisk, or is there not. How do i proceed, if there is nothing more to configure in stun, does that mean i can start configuring my clinets (xten and sipura) to use stun server? -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com http://www.axvoice.com http://www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com http://www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use stun server?
which stun server do you use? On 8/1/07, SIP [EMAIL PROTECTED] wrote: No... there's no STUN server built into Asterisk. Asterisk handles NAT in a different way... and is an endpoint rather than a proxy, so it doesn't really NEED STUN built into it. However, we run a STUN server on the same machine as an Asterisk server and see nothing in terms of load increase. STUN's footprint is rather negligible. N. Rizwan Hisham wrote: Ok thanx. One more thing to ask is: does asterisk has a stun server implemented in it or not. i mean does asterisk contain a stun server and provides any application which can be used for enabling the stun server in asterisk? On 8/1/07, *SIP* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: STUN is a pretty simplistic server. There's nothing else that needs to be configured on the STUN server side. It's pretty much either running or it's not. Just start plugging in the server to your clients and give it a whirl. It should work. N. Rizwan Hisham wrote: Hi all, This is the first time i am using stun with asterisk for nat problems. I have read the rfc which describes how stun works. i didnt have any problems understanding it. I have also intalled the stun server called stund which i downloaded from sourceforge. I have seen on the list that most people use stund here. I have started the stun server and its running silently. Now i dont know what to do next. Is there something more to it, like configuration files which i can use for special configuration for asterisk, or is there not. How do i proceed, if there is nothing more to configure in stun, does that mean i can start configuring my clinets (xten and sipura) to use stun server? -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com http://www.axvoice.com http://www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com http://www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
