<--- SIP read from UDP://10.52.23.100:5061 --->
INVITE sip:service@10.52.23.99:5060 SIP/2.0
Via: SIP/2.0/UDP 10.52.23.100:5061;branch=z9hG4bK-17481-74894-0
From: sipp <sip:sipp@10.52.23.100:5061>;tag=17481SIPpTag0074894
To: sut <sip:service@10.52.23.99:5060>
Call-ID: 74894-17481@10.52.23.100
CSeq: 1 INVITE
Contact: sip:sipp@10.52.23.100:5061
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length:   133

v=0
o=user1 53655765 2353687637 IN IP4 10.52.23.100
s=-
c=IN IP4 10.52.23.100
t=0 0
m=audio 6000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
<------------->
--- (11 headers 7 lines) ---
Sending to 10.52.23.100 : 5061 (no NAT)
Using INVITE request as basis request - 74894-17481@10.52.23.100
No user 'sipp' in SIP users list
Found peer 'sipp' for 'sipp' from 10.52.23.100:5061
Found RTP audio format 0
Peer audio RTP is at port 10.52.23.100:6000
Found audio description format PCMU for ID 0
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 10.52.23.100:6000
Looking for service in default (domain 10.52.23.99)
list_route: hop: <sip:sipp@10.52.23.100:5061>

<--- Transmitting (no NAT) to 10.52.23.100:5061 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.52.23.100:5061;branch=z9hG4bK-17481-74894-0;received=10.52.23.100
From: sipp <sip:sipp@10.52.23.100:5061>;tag=17481SIPpTag0074894
To: sut <sip:service@10.52.23.99:5060>
Call-ID: 74894-17481@10.52.23.100
CSeq: 1 INVITE
User-Agent: Asterisk PBX SVN-branch-1.6.0-r141567-/trunk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:service@10.52.23.99>
Content-Length: 0


<------------>
mediagateway3*CLI> Audio is at 10.52.23.99 port 11092
mediagateway3*CLI> Adding codec 0x4 (ulaw) to SDP
mediagateway3*CLI> 
<--- Transmitting (no NAT) to 10.52.23.100:5061 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.52.23.100:5061;branch=z9hG4bK-17481-74894-0;received=10.52.23.100
From: sipp <sip:sipp@10.52.23.100:5061>;tag=17481SIPpTag0074894
To: sut <sip:service@10.52.23.99:5060>;tag=as0ca1fc81
Call-ID: 74894-17481@10.52.23.100
CSeq: 1 INVITE
User-Agent: Asterisk PBX SVN-branch-1.6.0-r141567-/trunk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:service@10.52.23.99>
Content-Type: application/sdp
Content-Length: 229

v=0
o=root 1853459403 1853459403 IN IP4 10.52.23.99
s=Asterisk PBX SVN-branch-1.6.0-r141567-/trunk
c=IN IP4 10.52.23.99
t=0 0
m=audio 11092 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv



After that:

mediagateway3*CLI>
  * SIP Call
  Curr. trans. direction:  Incoming
  Call-ID:                74894-17481@10.52.23.100
  Owner channel ID:       SIP/5061-b7cf8b00
  Our Codec Capability:   524302
  Non-Codec Capability (DTMF):   1
  Their Codec Capability:   4
  Joint Codec Capability:   4
  Format:                 0x4 (ulaw)
  T.38 support            No
  Video support           No
  MaxCallBR:              384 kbps
  Theoretical Address:    10.52.23.100:5061
  Received Address:       10.52.23.100:5061
  SIP Transfer mode:      open
  NAT Support:            RFC3581
  Audio IP:               10.52.23.99 (local)
  Our Tag:                as0ca1fc81
  Their Tag:              17481SIPpTag0074894
  SIP User agent:
  Peername:               sipp
  Original uri:           sip:sipp@10.52.23.100:5061
  Caller-ID:              sipp
  Need Destroy:           No
  Last Message:           Rx: INVITE
  Promiscuous Redir:      No
  Route:                  sip:sipp@10.52.23.100:5061
  DTMF Mode:              rfc2833
  SIP Options:            (none)
  Session-Timer:          Inactive