On Mon, Oct 5, 2009 at 2:42 PM, Dale Worley <[email protected]> wrote: > On Mon, 2009-10-05 at 12:51 -0400, Jiann-Ming Su wrote: >> We're running into an issue with phone calls between sipX and Avaya >> terminate after 30 minutes. My colleague has observed that the Avaya >> will refresh the session according to its "Preferred Minimum Session >> Refresh Interval." Avaya sends an Invite packet to sipX using >> TCP/5060 (packet 50 below). sipX responds with a Trying 100 packet on >> UDP 5060 (packet 52). Avaya doesn't support SIP over UDP. The call >> terminates because the SIP conversation doesn't take place properly. > > A likely culprit is that the topmost Via header in the INVITE that the > Avaya sends does not specify that the transport must be TCP. (Or that > sipXecs is not respecting that Via header.) In any case, you need to > capture the entire contents of that INVITE message. >
Here's the INVITE header: 802.1Q Virtual LAN, PRI: 6, CFI: 0, ID: 1478 Internet Protocol, Src: 10.133.65.134 (10.133.65.134), Dst: 10.133.52.23 (10.133.52.23) Transmission Control Protocol, Src Port: 11369 (11369), Dst Port: 5060 (5060), Seq: 2404517912, Ack: 1245597638, Len: 39 [Reassembled TCP Segments (1279 bytes): #31(248), #33(248), #35(248), #37(248), #39(248), #41(39)] Session Initiation Protocol Request-Line: INVITE sip:[email protected]:61154;rinstance=d3c7beaddcffc9ac;x-sipX-nonat SIP/2.0 Method: INVITE [Resent Packet: False] Message Header From: "Lastname,First" <sip:[email protected]>;tag=0aa62f078aede1173f4ac32b4600 To: "10995" <sip:[email protected]>;tag=956e8f7f Call-ID: 0aa62f078aede1183f4ac32b4600 CSeq: 2 INVITE Max-Forwards: 70 Route: <sip:10.133.52.23:5060;lr;sipXecs-rs=%2Aauth%7E.%2Afrom%7EMGFhNjJmMDc4YWVkZTExNzNmNGFjMzJiNDYwMA%60%60%21d78d9e6cfed206e0e9dd4aa8ea1803f5> Via: SIP/2.0/UDP 10.133.65.134;branch=z9hG4bK80904b3a79aede1213f4ac32b4600 User-Agent: Avaya CM/R015x.02.0.947.3 Supported: timer, replaces, join, histinfo, 100rel Allow: INVITE, CANCEL, BYE, ACK, PRACK, SUBSCRIBE, NOTIFY, REFER, OPTIONS, INFO, PUBLISH Contact: "Lastname,First" <sip:[email protected];transport=tcp> Session-Expires: 240;refresher=uac Min-SE: 240 P-Asserted-Identity: "Lastname,First" <sip:[email protected]> Accept-Language: en P-Charging-Vector: icid-value="AAS:10530-f062aa001deae78c34a3f16462b" Content-Type: application/sdp Content-Length: 226 Message Body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 1 1 IN IP4 10.133.65.134 Session Name (s): - Connection Information (c): IN IP4 10.133.65.140 Bandwidth Information (b): AS:64 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 3748 RTP/AVP 0 18 127 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:18 G729/8000 Media Attribute (a): fmtp:18 annexb=no Media Attribute (a): rtpmap:127 telephone-event/8000 Media Attribute (a): ptime:10 "transport=tcp" is specified. We'll try to do a more comprehensive debug with sipviewer as Scott Lawrence suggested as well. -- Jiann-Ming Su "I have to decide between two equally frightening options. If I wanted to do that, I'd vote." --Duckman "The system's broke, Hank. The election baby has peed in the bath water. You got to throw 'em both out." --Dale Gribble "Those who vote decide nothing. Those who count the votes decide everything.” --Joseph Stalin _______________________________________________ sipx-dev mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-dev Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-dev sipXecs IP PBX -- http://www.sipfoundry.org/
