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@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; }; _______________________________________________ Serusers mailing list Serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers