[Remote SIP client] Couple of questions
Hello, I have a couple of questions related to having SIP users connecting from the Net to a Freeswitch server through NAT routers on both ends: 1. How must I configure routers on both ends? I understand that I need to route incoming TCP/UDP 5080 into the Freeswitch server, but what about the other router? I guess I also need to route this port to let the SIP phone ring, but what about data (RTP/RTCP)? 2. The Freeswitch server is connected to the POTS with either an OpenVox PCI card or a Linksys 3102 box: When a call is made between the POTS and a remote SIP phone (ie. out there on the Net, not on the same LAN as the Freeswitch server), is there a way for data to flow directly from the POTS to the remote SIP client instead of through the Freeswitch server? Thank you.
Just following up... did you get these questions ironed out? -MC On Wed, Mar 25, 2009 at 2:51 AM, Gilles <codecomplete@free.fr> wrote:
Hello,
I have a couple of questions related to having SIP users connecting from the Net to a Freeswitch server through NAT routers on both ends:
1. How must I configure routers on both ends? I understand that I need to route incoming TCP/UDP 5080 into the Freeswitch server, but what about the other router? I guess I also need to route this port to let the SIP phone ring, but what about data (RTP/RTCP)?
2. The Freeswitch server is connected to the POTS with either an OpenVox PCI card or a Linksys 3102 box: When a call is made between the POTS and a remote SIP phone (ie. out there on the Net, not on the same LAN as the Freeswitch server), is there a way for data to flow directly from the POTS to the remote SIP client instead of through the Freeswitch server?
Thank you.
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mercutioviz wrote:
Just following up... did you get these questions ironed out?
Not yet, but I do need to have a clear understanding about how to set things up when NAT is involved, especially when remote SIP users are also behind a NAT router, and especially if they can't make any change to it (eg. staying in a hotel, or don't have the skills required to open up ports). Thank you. -- View this message in context: http://www.nabble.com/-Remote-SIP-client--Couple-of-questions-tp22698296p230... Sent from the Freeswitch-users mailing list archive at Nabble.com.
definitely stop by IRC and talk real-time with others who've dealt with this kind of thing. -MC On Thu, Apr 16, 2009 at 7:44 AM, Fred-145 <codecomplete@free.fr> wrote:
mercutioviz wrote:
Just following up... did you get these questions ironed out?
Not yet, but I do need to have a clear understanding about how to set things up when NAT is involved, especially when remote SIP users are also behind a NAT router, and especially if they can't make any change to it (eg. staying in a hotel, or don't have the skills required to open up ports).
Thank you. -- View this message in context: http://www.nabble.com/-Remote-SIP-client--Couple-of-questions-tp22698296p230... Sent from the Freeswitch-users mailing list archive at Nabble.com.
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mercutioviz wrote:
definitely stop by IRC and talk real-time with others who've dealt with this kind of thing.
Thanks for the tip, but since this firewalls are a frequent issue with SIP, I think it'd be very useful if someone who has good experience with this wrote an article in the FS wiki for everyone to benefit. For instance, at this point, I don't know how data (SIP and RTP) flows between a remote SIP phone behind a NAT router, and a VoIP gateway located in the same LAN as a Freeswitch server both behind a NAT router. Do RTP packets go directly from the gateway to the remote SIP phone, or do they go through the FS server? That sort of thing :-) -- View this message in context: http://www.nabble.com/-Remote-SIP-client--Couple-of-questions-tp22698296p230... Sent from the Freeswitch-users mailing list archive at Nabble.com.
participants (3)
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Fred-145
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Gilles
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Michael Collins