Hi,all
This is my mode uac---sip server---uas.
When the UAC sipp sends INVITE, it goes through my SIP Server and the SIP
Server sends it to
the UAS sipp. However, the UAS doesn't see the INVITE ,the screen information
is
UAC:
Messages Retrans Timeout Unexpected-Msg
INVITE ----------> 1 0 0
100 <---------- 1 0 0 0
180 <---------- 1 0 0 0
200 <---------- 0 0 0 1
ACK ----------> 0 0
Pause [ 0ms] 0 0
BYE ----------> 0 0 0
200 <---------- 0 0 0 0
UAS:
Messages Retrans Timeout Unexpected-Msg
----------> INVITE 0 0 0 0
<---------- 180 0 0
<---------- 200 0 0 0
----------> ACK E-RTD1 0 0 0 0
----------> BYE 0 0 0 0
<---------- 200 0 0
------------------------------ Sipp Server Mode -------------------------------
I've seen on Wireshark that the INVITEs are really sent to the UAS sipp
address/port. What can be the cause of ignoring them?
That is my xml files:
Uac xml file:
<scenario name="calling">
<send retrans="500">
<![CDATA[
INVITE sip:[EMAIL PROTECTED];transport=udp SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch];rport
Max-Forwards: 70
Contact: <sip:[EMAIL PROTECTED]:[local_port];transport=[transport]>
To: <sip:[EMAIL PROTECTED]:[remote_port]>
From: "[field0]"<sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]
Call-ID: [call_id]
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: eventlist
User-Agent: eyeBeam release 3015c stamp 27107
Content-Length: 349
v=0
o=- 30972963 30973172 IN IP[media_ip_type] [media_ip]
s=eyeBeam
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 100 6 0 8 3 18 98 97 5 102 101
a=alt:1 1 : 367818CA DE47B706 [media_ip] [media_port]
a=fmtp:101 0-15
a=rtpmap:100 speex/16000
a=rtpmap:98 ilbc/8000
a=rtpmap:97 speex/8000
a=rtpmap:102 l16/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv
]]>
</send>
<recv response="100" >
</recv>
<recv response="180" >
</recv>
<recv response="200" >
</recv>
<send>
<![CDATA[
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch];rport
Max-Forwards: 70
Contact: <sip:[EMAIL PROTECTED]:[local_port];transport=[transport]>
To: <sip:[EMAIL PROTECTED]:[remote_port]>;tag=[call_number]
From: "[field0]"<sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]
Call-ID: [call_id]
CSeq: 1 ACK
User-Agent: eyeBeam release 3015c stamp 27107
Content-Length: [len]
]]>
</send>
<pause />
<send retrans="500">
<![CDATA[
BYE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch];rport
Max-Forwards: 70
Contact: <sip:[EMAIL PROTECTED]:[local_port];transport=[transport]>
To: <sip:[EMAIL PROTECTED]:[remote_port]>;tag=[call_number]
From: "[field0]"<sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]
Call-ID: [call_id]
CSeq: 2 BYE
User-Agent: eyeBeam release 3015c stamp 27107
Reason: SIP;description="User Hung Up"
Content-Length: [len]
]]>
</send>
<recv response="200" crlf="true">
</recv>
<!-- definition of the response time repartition table (unit is ms) -->
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
<!-- definition of the call length repartition table (unit is ms) -->
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>
UAS xml file
<scenario name="invite">
<recv request="INVITE" crlf="true">
</recv>
<send>
<![CDATA[
SIP/2.0 180 Ringing
Via:
SIP/2.0/[transport][remote_ip]:[remote_port];branch=[branch];rport=[remote_port]
Contact: <sip:[EMAIL PROTECTED]:[local_port];rinstance=6609d6a0ab91ecb6>
To: <sip:[EMAIL
PROTECTED]:[local_port];rinstance=6609d6a0ab91ecb6>;tag=[call_number]
From: "[field1]"<sip:[EMAIL PROTECTED]:[remote_port]>;tag=[call_number]
Call-ID: [last_Call-ID]
CSeq: 102 INVITE
User-Agent: eyeBeam release 3015c stamp 27107
Content-Length: [len]
]]>
</send>
<send retrans="500">
<![CDATA[
SIP/2.0 200 OK
Via:
SIP/2.0/[transport][remote_ip]:[remote_port];branch=[branch];rport=[remote_port]
Contact: <sip:[EMAIL PROTECTED]:[local_port];rinstance=6609d6a0ab91ecb6>
To: <sip:[EMAIL
PROTECTED]:[local_port];rinstance=6609d6a0ab91ecb6>;tag=[call_number]
From: "[field1]"<sip:[EMAIL PROTECTED]:[remote_port]>;tag=[call_number]
Call-ID: [last_Call-ID]
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: eventlist
User-Agent: eyeBeam release 3015c stamp 27107
Content-Length: [len]
v=0
o=- 28626111 28626185 IN IP[media_ip_type] [media_ip]
s=eyeBeam
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 18 101
a=alt:1 1 : 026EBD15 DE47B706 [media_ip] [media_port]
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
]]>
</send>
<recv request="ACK" optional="true" rtd="true" crlf="true" />
<recv request="BYE" />
<send>
<![CDATA[
SIP/2.0 200 OK
Via:
SIP/2.0/[transport][remote_ip]:[remote_port];branch=[branch];rport=[remote_port]
Contact: <sip:[local_ip]:[local_port];rinstance=6609d6a0ab91ecb6>
To: <sip:[EMAIL
PROTECTED]:[local_port];rinstance=6609d6a0ab91ecb6>;tag=[call_number]
From: "[field1]"<sip:[EMAIL PROTECTED]:[remote_port]>;tag=[call_number]
Call-ID: [last_Call-ID]
CSeq: 103 BYE
User-Agent: eyeBeam release 3015c stamp 27107
Content-Length: 0
]]>
</send>
<!-- Keep the call open for a while in case the 200 is lost to be -->
<!-- able to retransmit it if we receive the BYE again. -->
<!-- definition of the response time repartition table (unit is ms) -->
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
<!-- definition of the call length repartition table (unit is ms) -->
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>
Can anyone help me?
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