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The first line allows to access port 5060 (SIP port) and
the ports 10000 to 10999 (RTP ports, see command
line argument to the SIP NAT gateway). The second
line forwards packet arriving at port 5060 to the port
5061, on which the SIP NAT gateway is expecting
packets (using –connect-port 5061).
However, the method of using the IP gateway has the
disadvantage that SIP messages that go to other port
that 5060 (which is perfectly legal) do not pass the
SIP NAT gateway properly. Therefore, we recommend
using outbound proxy in the phones instead.
Quality of Service
Telephone calls require a good quality of service
(QoS), meaning that the associated media UDP pack-
ets are transported with a higher priority than other
packets.
Reliable protocols like TCP ensure by packet repeating
that the content does not get lost. This can become a
problem, when the same line is used for receiving
emails and VoIP. The underlying transport mecha-
nisms need to know about the priority of the packets.
Many mechanisms have been proposed for controlling
and ensuring the QoS ranging from RSVP, MPLS, ATM
etc. Practically, most installations support DiffSrv and
can deliver good quality with this mechanism.
Linux supports the different mechanisms. The SIP NAT
gateway uses DiffSrv marking of RTP packets. How-
ever, it is important that the kernel was compiled with
the corresponding configuration.
The SIP NAT gateway does not check if the underlying
bandwidth is enough for transporting the media
stream. If you try to make two phone calls over a 128
kBit/s line (occupying more than kBit/s per media
stream with ulaw encoding), will result in a breaking
phone call.
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