On 3/2/11 9:32 AM, Spinov Evgeniy wrote:
Unfortunately ngrep is unavailable right now, cause network was
configured to use public IPs. May be I'll can do that on development
network later. Right now development network using public`s also.
I'll try to sort out ngrep anyway.
I was giving FAEI to INVITEs from UAC to Asterisk and FAIE to INVITEs
from Asterisks to UAC. Everything was good except destination UDP port
to UAC 1. It was different then the source. As result UAC 1 didn't
received backflow.
You say about wrong port for RTP or for SIP?
For SIP be sure you call force_rport(). For RTP try eventually the flag
'r' in in parameters of force_rtp_proxy().
Also, may be this will help: Kamailio was unable to identify that faulty
UAC 1 is behind the NAT. I've tried nat_uac_test("31"), however -
nothing, while SIP headers were containing NATed IPs.
By NATed ip you mean private class, like 10... or 192.168...? If yes,
that is strange, can you add debugger module with cfgtrace enabled to
see what lines in the config file are executed for that call? (this is
assuming you are using v3.1.x, if not add xlog() messages in the config
to be sure the nat handling part is executed).
Cheers,
Daniel
So during tests
I've just forced NAT always. Without that I didn't had audio at all.
While with it - one way audio with faulty UAC and normal call for all
others.
Also, on faulty UAC 1 I had to use STUN server, while all other clients
worked without it. After going Asterisks public and changing kamailio
configuration for it, STUN no longer needed anywhere.
Just assuming fact, that router has bad ALG implementation. Is there any
workaround for it, may be forcing destination ports to source ones?
On Wed, 2011-03-02 at 09:30 +0100, Daniel-Constantin Mierla wrote:
Hello,
one option might be a bad ALG implementation in the router.
Can you send a full ngrep of such case? You can obfuscate the IP
addresses, use different ones for each point in the network and leave
the ports. Seeing SIP headers and SDP can indicate the presence of an
ALG or something broken in config logic.
Also, what is the parameter you give to force_rtp_proxy(...)?
Cheers,
Daniel
On 3/2/11 8:38 AM, Spinov Evgeniy wrote:
May be I miss some important details? No suggestions?
Thank you.
Hello, all.
Using nathelper + rtpproxy for subj. Kamailio has public and private
network interfaces. Asterisk is only private. RTP Proxy is working in
bridge mode and relaying traffic from UAC to Asterisks.
Everything is working fine, except one configuration. When the client is
behind router ( a specific one, I do not have an access there to
check ), and this UAC is making a call to other public extension, which
is behind router, then RTP Proxy is relaying traffic to the caller,
using another UDP port, then the packets arrive.
For instance:
UAC 1 -> UAC 2
PUBLIC_IP:10> KAMAILIO_IP:5555
KAMAILIO_IP:5678> PUBLIC_IP:12
While for the UAC 2 it looks like:
PUBLIC_IP:20> KAMAILIO_IP:6767
KAMAILIO_IP:4564> PUBLIC_IP:20
The source and destination UDP ports are the same. As result, I can hear
UAC 1 and he cannot hear me.
In case of we have UAC 3, which is behind other router, call is working
fine with same configuration.
"It's routers fault" you can say, but in the same configuration ( I mean
network, not kamailio ) it worked, but when RTPProxy was not in bridge
mode and Kamailio and Asterisks were in public network. Reinvites are
not allowed in both cases.
The question is, why the source and destination UDP ports are different?
Using STUN in first case, cause without it, private IP written in
contacts and as result, traffic relayed from Kamailio is incorrect,
cause heading to private network which is unreachable.
Any ideas where to dig?
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Daniel-Constantin Mierla
http://www.asipto.com
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