[Serusers] on hold problem
Greger V. Teigre
greger at teigre.com
Mon Oct 30 12:43:28 CET 2006
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 at 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 at sip.xyz.com;user=phone>;tag=10f10cec586cbf45
> SIP from address: sip:88008 at sip.xyz.com
> SIP tag: 10f10cec586cbf45
> To: "test-gxp2000"
> <sip:88009 at sip.xyz.com;user=phone>;tag=c6bad1edef48137f
> SIP Display info: "test-gxp2000"
> SIP to address: sip:88009 at sip.xyz.com
> SIP tag: c6bad1edef48137f
> Contact: <sip:88008 at x.x.x.x70:1025>
> Contact Binding: <sip:88008 at x.x.x.x70:1025>
> URI: <sip:88008 at x.x.x.x70:1025>
> SIP contact address: sip:88008 at x.x.x.x70:1025
> Supported: replaces, timer
> Call-ID: 98176a3a3b1280d6 at 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 at lists.iptel.org
>>> http://lists.iptel.org/mailman/listinfo/serusers
>>>
>
>
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