[SR-Users] FW: Client site NAT negotiation with no STUN.

[Daniel Tryba]

On Friday 14 February 2014 16:31:20 Rob Moore wrote:
> Anyway, I've had a look at the change you've suggested, unfortunately it
> seems to have made little difference and the RTP proxy is still trying to
> send traffic to the client SIPPhones private ip address and not the
> firewall.
> 
> It's probably worth clarifying (seen as the diagram got mangled)  that I
> don't have an RTPproxy at our client site only in our data centre paid
> with our kamailio.

Well, we have the same setup:

U 109.235.34.226:42259 -> 109.235.32.42:5060
INVITE sip:0880100705 at pisco.pocos.nl SIP/2.0.
Via: SIP/2.0/UDP 10.0.3.175:5062;branch=z9hG4bK-54389de.
From: "tryba" <sip:tryba at pisco.pocos.nl>;tag=ea2052858ec448f9o2.
To: <sip:0880100705 at pisco.pocos.nl>.
Call-ID: 20654a75-2dca31c9 at 10.0.3.175.
CSeq: 101 INVITE.
Max-Forwards: 70.
Contact: "tryba" <sip:tryba at 10.0.3.175:5062>.
Expires: 240.
User-Agent: Linksys/SPA962-6.1.3(a).
Content-Length: 205.
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER.
Supported: replaces.
Content-Type: application/sdp.
.
v=0.
o=- 872251 872251 IN IP4 10.0.3.175.
s=-.
c=IN IP4 10.0.3.175.
t=0 0.
m=audio 16404 RTP/AVP 2 101.
a=rtpmap:2 G726-32/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:30.
a=sendrecv.

...

U 109.235.32.42:5060 -> 109.235.34.226:42259
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 
10.0.3.175:5062;rport=42259;received=109.235.34.226;branch=z9hG4bK-54389de.
Record-Route: 
<sip:109.235.32.42;lr=on;ftag=ea2052858ec448f9o2;vsf=AAAAAF9BSFZRc0RZQ1NfHjAfChwQQUAcb2Nvcy5ubA--;nat=yes;vsf=Q2lwOnRyeWJhQHBpc2NvLnBvY29zLm5s;nat=yes>.
From: "tryba" <sip:tryba at pisco.pocos.nl>;tag=ea2052858ec448f9o2.
To: <sip:0880100705 at pisco.pocos.nl>;tag=as15267783.
Call-ID: 20654a75-2dca31c9 at 10.0.3.175.
CSeq: 101 INVITE.
Server: Pocos.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces, timer.
Contact: <sip:+31880100705 at 109.235.32.xx>.
Content-Type: application/sdp.
Content-Length: 211.
.
v=0.
o=root 87313901 87313901 IN IP4 109.235.32.42.
s=Asterisk PBX 1.6.2.9-2+squeeze11.
c=IN IP4 109.235.32.42.
t=0 0.
m=audio 45940 RTP/AVP 2.
a=rtpmap:2 G726-32/8000.
a=ptime:20.
a=sendrecv.
a=nortpproxy:yes.


The natted client will start the RTP stream towards 109.235.32.42:45940 and 
after about 5-10 packets kamailio/rtpproxy will start sending to the source of 
the incoming steam.

Attached a flow from an other test call.

As far as I know the only change I made was moving the rtpproxy_manage() a 
bit.

Maybe I'll have some time monday to generate a small proof of concept config, 
if nobody gives the answer before that time.

-- 

POCOS B.V. - Croy 9c - 5653 LC Eindhoven
Telefoon: 040 293 8661 - Fax: 040 293 8658
http://www.pocos.nl/   - http://www.sipo.nl/
K.v.K. Eindhoven 17097024
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