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Archived from groups: comp.dcom.voice-over-ip (More info?)
This may shed some light for those who are trying to estabilish VOIP
behind a NAT router. - RM
from http://www.voip-info.org/wiki-NAT+and+VOIP
:
The Trouble with NAT and VOIP
"In addition, the way in which conventional VoIP protocols are designed
is also posing a problem to VoIP traffic passing through NAT.
Conventional VoIP protocols only deal with the signalling of a telephone
connection. The audio traffic is handled by another protocol and to make
matters worse, the port on which the audio traffic is sent is random.
The NAT router may be able to handle the signalling traffic, but it has
no way of knowing that the audio traffic is related to the signalling
and should hence be passed to the same device the signalling traffic is
passed to. As a result, the audio traffic is simply discarded.
"At first, for both the calling and the called party everything will
appear just fine. The called party will see the calling party's Caller
ID and the telephone will ring while the calling party will hear a
ringing feedback tone at the other end. When the called party picks up
the telephone, both the ringing and the associated ringing feedback tone
at the other end will stop as one would expect. However, the calling
party will not hear the called party (one way audio) and the called
party may not hear the calling party either (no audio).
"The issue of NAT Traversal is a major problem for the widespread
deployment of VOIP. Yet, the issue is non-trivial and there are no
simple solutions."
This may shed some light for those who are trying to estabilish VOIP
behind a NAT router. - RM
from http://www.voip-info.org/wiki-NAT+and+VOIP
:
The Trouble with NAT and VOIP
"In addition, the way in which conventional VoIP protocols are designed
is also posing a problem to VoIP traffic passing through NAT.
Conventional VoIP protocols only deal with the signalling of a telephone
connection. The audio traffic is handled by another protocol and to make
matters worse, the port on which the audio traffic is sent is random.
The NAT router may be able to handle the signalling traffic, but it has
no way of knowing that the audio traffic is related to the signalling
and should hence be passed to the same device the signalling traffic is
passed to. As a result, the audio traffic is simply discarded.
"At first, for both the calling and the called party everything will
appear just fine. The called party will see the calling party's Caller
ID and the telephone will ring while the calling party will hear a
ringing feedback tone at the other end. When the called party picks up
the telephone, both the ringing and the associated ringing feedback tone
at the other end will stop as one would expect. However, the calling
party will not hear the called party (one way audio) and the called
party may not hear the calling party either (no audio).
"The issue of NAT Traversal is a major problem for the widespread
deployment of VOIP. Yet, the issue is non-trivial and there are no
simple solutions."