Full NAT bypass solution with STUN/ICE/FS but without TURN: is possible?
Hi all, i need to setup a server that handle VoIP connection in a way that always works in the most efficient way respect to latency and NAT traversal. FS provide very good NAT detection system but unfortunately when it fail, it just fail and does not have any kind of fallback mechanism. If FS do bypass-media and the client cannot communicate because the phone call is lost. If FS do proxy-media, the clients can always communicate (over UDP) but there is added latency. So we cannot rely only on FS NAT detection system and need also some client side system. Now, in theory the best approach would be not to touch FS but just use client-side system based on ICE/STUN with a TURN server-side component as a fallback. However there's no widely available TURN server, or at least TURN is not widely and well diffused just now. I am wondering whether it may be possible to make a setup where: - Client try to establish peer to peer connectivity with STUN/ICE (so FS is doing bypass-media not touching the SDP) - If they cannot work, will go in fallback to FS (but with proxy-media) instead than going via TURN So using the FS B2BUA proxy-media feature as alternative to TURN server, but letting clients to dynamically try to connect directly one each other by using the ICE methods that are dynamic. Does it seems something feasible? Fabio Pietrosanti
Fabio, this sounds like a question to ask to your SIP phones vendor. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 08/06/2010 à 10:37, Fabio Pietrosanti (naif) a écrit :
Hi all,
i need to setup a server that handle VoIP connection in a way that always works in the most efficient way respect to latency and NAT traversal.
FS provide very good NAT detection system but unfortunately when it fail, it just fail and does not have any kind of fallback mechanism. If FS do bypass-media and the client cannot communicate because the phone call is lost. If FS do proxy-media, the clients can always communicate (over UDP) but there is added latency.
So we cannot rely only on FS NAT detection system and need also some client side system.
Now, in theory the best approach would be not to touch FS but just use client-side system based on ICE/STUN with a TURN server-side component as a fallback.
However there's no widely available TURN server, or at least TURN is not widely and well diffused just now.
I am wondering whether it may be possible to make a setup where: - Client try to establish peer to peer connectivity with STUN/ICE (so FS is doing bypass-media not touching the SDP) - If they cannot work, will go in fallback to FS (but with proxy- media) instead than going via TURN
So using the FS B2BUA proxy-media feature as alternative to TURN server, but letting clients to dynamically try to connect directly one each other by using the ICE methods that are dynamic.
Does it seems something feasible?
Fabio Pietrosanti
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I am the SIP phone vendor, but still did not have experience with NAT bypass technologies and so i am asking whether this is a setup that may be feasible or not from the "protocols" point of view and from FS point of view. I can install a TURN server but if possible i would like to avoid setting up server-side systems other than FS. Fabio On 08/06/10 10.53, David Ponzone wrote:
Fabio,
this sounds like a question to ask to your SIP phones vendor.
with freeswitch, in 99% of the situations, just using rport will pretty much work, stun is a bonus in a couple cases. This will get you through most nat. There are a few stuborn fools who it won't work with. But if they can not provide a working sip alg, then I think those devices are pretty much broken at that point, while not technically, in any practical sense. Mike On Jun 8, 2010, at 5:00 AM, Fabio Pietrosanti (naif) wrote:
I am the SIP phone vendor, but still did not have experience with NAT bypass technologies and so i am asking whether this is a setup that may be feasible or not from the "protocols" point of view and from FS point of view.
I can install a TURN server but if possible i would like to avoid setting up server-side systems other than FS.
Fabio
On 08/06/10 10.53, David Ponzone wrote:
Fabio,
this sounds like a question to ask to your SIP phones vendor.
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participants (3)
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David Ponzone
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Fabio Pietrosanti (naif)
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Michael Jerris