Ugo Bellavance wrote: > Ugo Bellavance wrote: >> Hi, >> >> On my linksys/sipura phones/ATA, there is a setting called "NAT >> Mapping Enable" and another called "NAT Keep Alive Enable" >> >> These settings must be on in my setup so that my phones/ATA remain >> connected to my * server. My setup is: >> >> Home LAN - Pfsense (NAT, Dynamic Public IP)- Internet - PFsense (1-to-1 >> NAT, Static public IP) - Asterisk server. >> >> I was wondering: >> >> What are doing those parameters? >> >> I looked on my Polycom 330 and I haven't found anything similar... Is >> Linksys the only Mfg that has a similar setting? The polycom doesn't >> have STUN settings either. I'm looking to buy some SNOM phones (M3 and >> a wired one), does SNOM phones have something similar? >> >> BTW, are there public STUN servers, or must I have my one to use it? >> >> Regards, >> >> Ugo > > I found a part of the answer here: > > http://www.sipura.com/Documents/SPA941AdminGuide.pdf, page 41, but I'm > still wondering how to get many polycoms working in a setup like mine... > Or Aastra, or maybe Snom.
Here is the text of the explanation: In the case of SIP, the addresses where messages/data should be sent to an SPA are embedded in the SIP messages sent by the device. If the SPA is sitting behind a NAT, the private IP address assigned to it is not usable for communications with the SIP entities outside the private network. The SPA must substitute the private IP address information with the proper external IP address/port in the mapping chosen by the underlying NAT to communicate with a particular public peer address/port. For this the SPA needs to perform the following tasks: -Discover the NAT mappings used to communicate with the peer. This could be done with the help of some external device. For example a server could be deployed on the external network such that the server will respond to a special NAT-Mapping-Discovery request by sending back a message to the source IP address/port of the request, where the message will contain the source IP address/port of the original request. The SPA can send such a request when it first attempts to communicate with a SIP entity in the public network and stores the mapping discovery results returned by the server. -Communicate the NAT mapping information to the external SIP entities. If the entity is a SIP Registrar, the information should be carried in the Contact header that overwrites the private address/port information. If the entity is another SIP UA when establishing a call, the information should be carried in the Contact header as well as in the SDP embedded in SIP message bodies. The VIA header in outbound SIP requests might also need to be substituted with the public address if the UAS relies on it to route back responses. Doesn't it look like STUN? _______________________________________________ --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
