30 Oct
2006
30 Oct
'06
1:43 p.m.
That's what the nat=yes is for. This means that you have a client that
does not keep the route parameter as it should.
Depending on which UA puts on hold (NATed or not), you may not be able
to use the NAT tests to detect NATed (because they are not).
A workaround would be to lookup("location") for reINVITEs.
g-)
Shaun Hofer wrote:
I found nat=yes was set for the first reINVITE to set
hold but not the second.
All the nat tests (tried all the modes for client_nat_test and nat_uac_test)
seem to fail picking up that the second reINVITE (take off hold INVITE) is
being nated, thus mediaproxy isn't used. From what I can see in the packet
captures and what the nat tests test for, they would fail to pick up the fact
that it is NAt'ed. (Packet Capture at the bottom)
At the monment I'm trying to figure a way to compare the owner field with
source ip address. These two IP address are different if NAT'ed. Owner is the
private IP address while the source IP has the NAT public IP address.
Been trying something like:
if(!search("o=[0-9]* [0-9]* [0-9]* IN IP4 \Q$src_ip\E$")) {
setflag(6);
use_media_proxy();
}
The problem I'm facing is it seems that search doesn't take in variables
($src_ip). The \Q and \E are used to quote thus avoid escaping the dots.
Is there way for search to take in variables ? Or better way to compare these
values?
Another idea that came to me: Would checking the ruri or Route for ser's
server ip address be better way, or is this abit dangerous to be
routing/mediaproxying according to this?
-Shaun
The Following is a Packet Capture of a packet sent to SER to take the phones
off hold (x.x.x.x3 = sip.xyz.com):
Session Initiation Protocol
Request-Line: INVITE sip:88009@x.x.x.x6:5516;user=phone SIP/2.0
Method: INVITE
Resent Packet: False
Message Header
Via: SIP/2.0/UDP x.x.x.x70:1025;branch=z9hG4bK2d93cdb6357ddb75
Route: <sip:x.x.x.x3;ftag=10f10cec586cbf45;lr=on>
From: <sip:88008@sip.xyz.com;user=phone>;tag=10f10cec586cbf45
SIP from address: sip:88008@sip.xyz.com
SIP tag: 10f10cec586cbf45
To: "test-gxp2000"
<sip:88009@sip.xyz.com;user=phone>;tag=c6bad1edef48137f
SIP Display info: "test-gxp2000"
SIP to address: sip:88009@sip.xyz.com
SIP tag: c6bad1edef48137f
Contact: <sip:88008@x.x.x.x70:1025>
Contact Binding: <sip:88008@x.x.x.x70:1025>
URI: <sip:88008@x.x.x.x70:1025>
SIP contact address: sip:88008@x.x.x.x70:1025
Supported: replaces, timer
Call-ID: 98176a3a3b1280d6(a)192.168.1.3
CSeq: 22284 INVITE
User-Agent: Grandstream GXP2000 1.1.0.16
Max-Forwards: 70
Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Content-Length: 267
Message body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): 88008 8000 8002 IN IP4 10.0.10.111
Owner Username: 88008
Session ID: 8000
Session Version: 8002
Owner Network Type: IN
Owner Address Type: IP4
Owner Address: 10.0.10.111
Session Name (s): SIP Call
Connection Information (c): IN IP4 x.x.x.x70
Connection Network Type: IN
Connection Address Type: IP4
Connection Address: x.x.x.x70
Time Description, active time (t): 0 0
Session Start Time: 0
Session Stop Time: 0
---snip----
On Wednesday 25 October 2006 17:14, Greger V. Teigre wrote:
- Make sure nat=yes is found in the Route set of
the reINVITE.
- Look at the mediaproxy messages in /var/log/messages, you should get
one for the INVITE and then one for the OK, but not '' empty response.
g-)
Shaun Hofer wrote:
> Hi,
>
> I've been having a problem, where audio is lost either in one or both
> directions when conversaion is taken off 'hold'. The parties are both
> behind NAT and depending UA as whether one or both loose audio. From what
> I can tell its to do with my loose route and nathelper, and how my
> ser.cfg deals with the take off hold INVITE from the phones. When the
> call is taken off hold the rtp streams aren't setup properly again (eg
> not using mediaproxy correctly). What is the best way to solve this
> problem?
>
> I've seen similarly posts to the mailing list about this problem with no
> solution posted.
> http://lists.iptel.org/pipermail/serusers/2006-March/027424.html
> http://lists.iptel.org/pipermail/serusers/2006-April/027885.html
> http://lists.iptel.org/pipermail/serusers/2006-May/028407.html
>
> Thanks
> Shaun
>
>
> I have a similarly config to getting started guides ser.cfg
>
> # -----------------------------------------------------------------
> # Loose Route Section
> # -----------------------------------------------------------------
>
> if (loose_route()) {
> if (!has_totag()) {
> sl_send_reply("403", "Forbidden");
> break;
> };
> if (method=="INVITE") {
> if ((method=="INVITE" ||
method=="REFER")
> && !has_totag()) {
> if
(!proxy_authorize("","subscriber")) {
> proxy_challenge("","0");
> break;
> } else if (!check_from()) {
> sl_send_reply("403", "Use
> From=ID"); break;
> };
> consume_credentials();
> };
> if
> (client_nat_test("3")||search("^Route:.*;nat=yes")) {
> setflag(6);
> use_media_proxy();
> };
> };
> route(1);
> break;
> };
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