Hi all,
I have tested successfully Softphone as client to connect Asterisk as Server.
I want to test Sipp as client to connect Asterisk as Server, because I
want to test with more and more users.
I have tried to create a scenario but it doesn't create a session.
In Asterisk, it returns message : 488 Not Acceptable here. I don't
understand this reason, because SDP in message INVITE, I copy from
message of Softphone.
I have attached my scenario,
Thank in advance for all helps!
Best regards,
T.Q.Tuan
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<scenario name="Session for conference">
<send retrans="500">
<![CDATA[
REGISTER sip:[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
Max-Forwards: 20
From: "[field0]" <sip:[fiel...@[field2]>;tag=[call_number]
To: "[field0]" <sip:[fiel...@[field2]>
P-Access-Network-Info: 3GPP-UTRAN-TDD;utran-cell-id-3gpp=C359A3913B20E
Call-ID: [call_id]
CSeq: 1 REGISTER
Contact: <sip:[fiel...@[local_ip]:[local_port]>
Expires: 7200
Content-Length: [len]
User-Agent: Sipp v1.1-TLS, version 20061124
Authorization: Digest username="[fiel...@[field2]", realm="[field2]"
Supported: path
]]>
</send>
<recv response="401" auth="true">
</recv>
<send retrans="500">
<![CDATA[
REGISTER sip:[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
Route: [$1]
Max-Forwards: 20
From: "[field0]" <sip:[fiel...@[field2]>;tag=[call_number]
To: "[field0]" <sip:[fiel...@[field2]>
P-Access-Network-Info: 3GPP-UTRAN-TDD;utran-cell-id-3gpp=C359A3913B20E
Call-ID: [call_id]
CSeq: 2 REGISTER
Contact: <sip:[fiel...@[local_ip]:[local_port]>
Expires: 7200
Content-Length: 0
User-Agent: Sipp v1.1-TLS, version 20061124
[field3]
Supported: path
]]>
</send>
<recv response="200">
<action>
<ereg regexp=".*" search_in="hdr" header="Service-Route:" assign_to="1" />
</action>
</recv>
<pause milliseconds="500" crlf="true" />
<send retrans="500">
<![CDATA[
INVITE sip:[servi...@[remote_ip]:5060 SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
Max-Forwards: 20
Route: <sip:pcscf.open-ims.test:4060;lr>,[$1]
P-Preferred-Identity: <sip:[fiel...@[field2]>
Privacy: none
P-Access-Network-Info: 3GPP-UTRAN-TDD;utran-cell-id-3gpp=C359A3913B20E
From: <sip:[fiel...@[field2]>;tag=[call_number]
To: <sip:[servi...@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: [cseq] INVITE
Contact: <sip:[fiel...@[local_ip]:[local_port]>
Expires: 7200
User-Agent: Sipp v1.1-TLS, version 20061124
Authorization: Digest username="[field1]", realm="[field2]"
Allow: ACK, BYE, CANCEL, INVITE, REFER, OPTIONS, INFO, REGISTER, NOTIFY, UPDATE, SUBSCRIBE, PRACK
Content-Type: application/sdp
Content-Length: [len]
v=0
o=- 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=SJphone
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio 5062 RTP/AVP 113 0 8 9 101 120
a=senddrecv
a=rtpmap:113 Speex/16000/1
a=fmtp:113 sr=16000,mode=any
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:9 G722/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32,36
a=rtpmap:120 NSE/8000
a=fmtp:120 192-193
m=video 5064 RTP/AVP 99 31
b=AS:4096
b=TIAS:4096000
a=sendrecv
a=rtpmap:99 theora/90000
a=fmtp:99 height=576;width=704
a=rtpmap:31 h263/90000
a=fmtp:31 CIF=1;QCIF=1
]]>
</send>
<recv response="401" auth="true">
</recv>
<send retrans="500">
<![CDATA[
INVITE sip:[servi...@[remote_ip]:5060 SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
Max-Forwards: 20
Route: <sip:pcscf.open-ims.test:4060;lr>,[$1]
P-Preferred-Identity: <sip:[fiel...@[field2]>
Privacy: none
P-Access-Network-Info: 3GPP-UTRAN-TDD;utran-cell-id-3gpp=C359A3913B20E
From: <sip:[fiel...@[field2]>;tag=[call_number]
To: <sip:[servi...@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: [cseq] INVITE
Contact: <sip:[fiel...@[local_ip]:[local_port]>
Expires: 7200
User-Agent: Sipp v1.1-TLS, version 20061124
[field3]
Allow: ACK, BYE, CANCEL, INVITE, REFER, OPTIONS, INFO, REGISTER, NOTIFY, UPDATE, SUBSCRIBE, PRACK
Content-Type: application/sdp
Content-Length: [len]
v=0 -53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=Opal SIP Session
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio 5062 RTP/AVP 113 0 8 9 101 120
a=senddrecv
a=rtpmap:113 Speex/16000/1
a=fmtp:113 sr=16000,mode=any
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:9 G722/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32,36
a=rtpmap:120 NSE/8000
a=fmtp:120 192-193
m=video 5064 RTP/AVP 99 31
b=AS:4096
b=TIAS:4096000
a=sendrecv
a=rtpmap:99 theora/90000
a=fmtp:99 height=576;width=704
a=rtpmap:31 h263/90000
a=fmtp:31 CIF=1;QCIF=1
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="180" optional="true">
</recv>
<recv response="403" optional="true" next="1">
</recv>
<recv response="404" optional="true" next="1">
</recv>
<recv response="408" optional="true" next="1">
</recv>
<recv response="200" rrs="true">
</recv>
<send crlf="true" >
<![CDATA[
ACK sip:[servi...@[local_ip]:5060 SIP/2.0
[last_Via:]
Max-Forwards: 20
[routes]
From: <sip:[fiel...@[field2]>;tag=[call_number]
[last_To:]
Call-ID: [call_id]
CSeq: [cseq] ACK
Content-Length: 0
]]>
</send>
<!-- Play a pre-recorded PCAP file (RTP stream) -->
<nop>
<action>
<exec play_pcap_audio="8Khz.pcap"/>
</action>
</nop>
<nop>
<action>
<exec play_pcap_video="paris.pcap"/>
</action>
</nop>
<pause milliseconds="60000" crlf="true" />
<!--
<send retrans="500">
<![CDATA[
REGISTER sip:[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
Max-Forwards: 20
From: "[field0]" <sip:[fiel...@[field2]>;tag=[call_number]
To: "[field0]" <sip:[fiel...@[field2]>
P-Access-Network-Info: 3GPP-UTRAN-TDD;utran-cell-id-3gpp=C359A3913B20E
Call-ID: [call_id]
CSeq: 3 REGISTER
Contact: <sip:[fiel...@[local_ip]:[local_port]>
Expires: 0
Content-Length: 0
User-Agent: Sipp v1.1-TLS, version 20061124
Authorization: Digest username="[fiel...@[field2]", realm="[field2]"
Supported: path
]]>
</send>
<recv response="401" auth="true">
</recv>
<send retrans="500">
<![CDATA[
REGISTER sip:[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
Route: [$1]
Max-Forwards: 20
From: "[field0]" <sip:[fiel...@[field2]>;tag=[call_number]
To: "[field0]" <sip:[fiel...@[field2]>
P-Access-Network-Info: 3GPP-UTRAN-TDD;utran-cell-id-3gpp=C359A3913B20E
Call-ID: [call_id]
CSeq: 4 REGISTER
Contact: <sip:[fiel...@[local_ip]:[local_port]>
Expires: 0
Content-Length: 0
User-Agent: Sipp v1.1-TLS, version 20061124
[field3]
Supported: path
]]>
</send>
<recv response="200" optional= "true" >
</recv>
<recv request="BYE">
</recv>
<send retrans="500" crlf="true">
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<recv response="200" rrs="true">
</recv>
-->
<pause milliseconds="5000" crlf="true" />
<send retrans="500">
<![CDATA[
BYE sip:157.159.16.91:5060 SIP/2.0
[last_Via:]
Max-Forwards: 20
[routes]
From: "[field0]" <sip:[fiel...@[field2]>;tag=[call_number]
To: <sip:[servi...@[remote_ip]:5060>
P-Preferred-Identity: <sip:[fiel...@[field2]>
P-Access-Network-Info: 3GPP-UTRAN-TDD;utran-cell-id-3gpp=C359A3913B20E
Call-ID: [call_id]
CSeq: [cseq] BYE
Content-Length: 0
]]>
</send>
<label id="1"/>
<label id="2"/>
<!-- 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>
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