Re: [Serusers] on hold problem

Hi, * Greger V. Teigre <greger(a)teigre.com> [061030 12:44]: This could cause intresting other issues. If a user has 2 phones registerd. the reinvite might get forked to both phones, wich (in a case I saw) made one phone ring, while the other picked up the call.

-A > 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; > >>> }; > >>>_______________________________________________ > >>>Serusers mailing list > &gt;&gt;&gt;Serusers(a)lists.iptel.org > >>>http://lists.iptel.org/mailman/listinfo/serusers > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers