Oh, ok, my RTP-Proxies do just this, so it would work anyways (and
that's also the behaviour which is necessary for some T.38
implementations).
I was just thinking about this situation and thought that there could be
an issue. False alarm :o)
Andreas
Klaus Darilion wrote:
Andreas Granig wrote:
Klaus,
Another, not directly related question:
If you have a nated UAC, RTP-Proxies usually wait for packets arriving
on each call leg before bridging them together. Now if you have a
peering with another provider which has the same setup using i-enum
and peering policies, you are stuck because both RTP-Proxies wait for
packets to arrive from the peering-leg. How would you handle that? Is
there an option in the peering policies, which negotiates which
provider's RTP-proxy is going to be used?
The RTP proxy should wait for packets from both destinations. It should
forward immediately to the socket announced in the SDP and fall back to
the socket from which it received the packet. If the both RTP proxies
have a public IP then the socket in the SDP can be reached directly.
I think rtpproxy will forward to the IP:port in the SDP and then fall
back to the IP:port a packet is received from if this is different.
I do not know how mediaproxy handles this.
In a "strict" peering environment you do not want your rtp proxy to
change the destination if someone sends you an RTP packet from any
destiantion. Thus you will tell the rtp proxy to by symmetric on one
side and to use the SDP on the other side. (like the "r" flag
http://www.openser.org/docs/modules/1.1.x/nathelper#AEN275)
regards
klaus
Andreas
Klaus Darilion wrote:
Probably not solving your problem but this is my newest pragmatic
aproach:
A client should support symmetric SIP. Thus, I use force_rport for all
local clients. As usually also all SIP proxies are symmetric I also do
force_port for requests from external nodes.
For REGISTER I do not trust the information in the Contact header at
all -
I always use fix_nated_register. Further, I always use fix_nated_contact
for local SIP users - thus for SIP NAT traversal I do not need any
tests.
Regarding RTP NAT traversal - if you want to save bandwidth on your RTP
proxy - of course you still need a nat-test.
regards
klaus
On Tue, February 13, 2007 17:36, Andreas Granig said:
Hi,
Today I found a UAC which is *not* located behind NAT (public IP
1.2.3.4) and sends this Via-Header, which seems perfectly valid
according to RFC3261:
SIP/2.0/UDP VINNASUP06C:5060;maddr=1.2.3.4;branch=z9hG4bK-2198d2
I used to check for nated clients using nat_uac_test("3"), which
detects
NAT in this case, because the host-part doesn't match the
received-address. So is the test-flag "2" useless, since the host-part
can be "hostname / IPv4address / IPv6reference", or should this
particular test be extended to also check for the maddr-parameter?
In the meanwhile, I've changed my nat-test to "17" for only testing
Contact and Via-Port instead of Contact and Via-Address, but it's still
not optimal.
Any opinions on this?
Andreas
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