09-19-2013 06:29 AM - edited 03-16-2019 07:27 PM
How does CUSP decide when to use TCP vs UDP when forwarding INVITES to CUCM when both UDP and TCP are configured on the Network Listen Ports? We have two different calls flows coming from our SBC to CUSP. Both INVITES have UDP as the transport. CUSP forwards one call flow to CUCM with TCP as the transport and the other as UDP. Is there something unique to the first INVITE below that would signal CUSP to use TCP?
Invite from SBC that CUSP adds TCP as the transport in the VIA header towards CUCM
40:13.979 On [2:245]10.XXX.XXX.XXX:5060 sent to 10.240.XXX.X:5060
INVITE sip:XXXXXXXXXX@cusplab:5060 SIP/2.0
Via: SIP/2.0/UDP 10.XXX.XXX.XXX:5060;branch=z9hG4bKm2u8f32038e12is8p3v1.1
From: "Test User 1" <sip:5045824230@12.194.11.53:5060>;tag=SDi82if02-13984769016573306_c2b04.1.2.1378274120440.0_1121221_2220804
To: <sip:+1XXXXXXXXXX@32.252.49.186>
Call-ID: SDi82if02-2664bc8b12611edfd30a0552529e0b90-cggngq0
CSeq: 2 INVITE
Session-Expires: 1800
Min-SE: 1800
Allow-Events: telephone-event
Cisco-Guid: 3327242752-0000065536-0000181788-1640976394
Acme-Call-ID: 9AA7C362-1BBC11E3-81019B0F-C8BF5371
Timestamp: 1379084661
Expires: 180
Supported: timer,replaces,sdp-anat
P-Asserted-Identity: "Test User 1" <sip:5045824230@12.194.11.53>
Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 59
Contact: <sip:10.XXX.XXX.XXX:5060;transport=udp>
Content-Length: 325
Content-Disposition: session; handling=required
Content-Type: application/sdp
v=0
o=CiscoSystemsSIP-GW-UserAgent 2487 3295 IN IP4 10.XXX.XXX.XXX
s=SIP Call
c=IN IP4 10.XXX.XXX.XXX
t=0 0
m=audio 55178 RTP/AVP 18 100 101
c=IN IP4 10.XXX.XXX.XXX
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
INVITE from SBC that CUSP adds UDP as the transport in the VIA header
15:14.881 On [2:245]10.XXX.XXX.XXX:5060 sent to 10.240.XXX.X:5060
INVITE sip:XXXXXXXXXX@cusplab:5060 SIP/2.0
Via: SIP/2.0/UDP 10.XXX.XXX.XXX:5060;branch=z9hG4bKg04f5a105040ei4764o1.1
From: "Test User 2" <sip:XXXXXXXXXX@12.194.137.181:5060>;tag=SDdeh1202-125657148404519E-4_c2b09.2.1.1376631876821.0_2507555_4963634
To: <sip:XXXXXXXXX@32.252.49.186>
Call-ID: SDdeh1202-c8199578035cb96793dbe0fb6ef98d84-cggjoq2
CSeq: 2 INVITE
P-Asserted-Identity: "WIRELESS CALLER" <sip:XXXXXXXXXX@12.194.137.181:5060>
Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Max-Forwards: 64
Contact: <sip:10.XXX.XXX.XXX:5060;transport=udp>
Content-Length: 266
Content-Disposition: session; handling=required
Content-Type: application/sdp
v=0
o=Sonus_UAC 12183 15514 IN IP4 10.XXX.XXX.XXX
s=SIP Media Capabilities
c=IN IP4 10.XXX.XXX.XXX
t=0 0
m=audio 55134 RTP/AVP 18 0 100
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=sendrecv
a=maxptime:30