On Mon, 2009-10-05 at 15:15 -0400, Jiann-Ming Su wrote:
> 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
As you can see, the topmost Via header is:
Via: SIP/2.0/UDP
10.133.65.134;branch=z9hG4bK80904b3a79aede1213f4ac32b4600
That instructs the recipient to send the response via *UDP* to IP
address 10.133.65.134, port 5060 (default for UDP). See RFC 3261
sections 8.1.1.7, 20.42, etc.
It is true that the Contact header specifies "transport=tcp", and as a
result, any *requests* sent *from* sipXecs *to* the Avaya for this
dialog will be sent using TCP. But the Contact header does not control
how responses are to be sent.
Dale
_______________________________________________
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/
