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scenarios/invite_voicebox: invite goes to voicebox

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vseva/3.0
Victor Seva 13 years ago
parent 8110096f4a
commit ab2eadc426
10 changed files with 543 additions and 0 deletions
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32
scenarios/invite_voicebox/0001_test.yml.tt2
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# REGISTER scenario
flow:
- start|MAIN:
- start|ROUTE_NET_INFO:
- end|ROUTE_NET_INFO:
- start|ROUTE_PRX_REQUEST:
- start|ROUTE_INVITE:
- start|ROUTE_LOAD_CALLEE_DOMAIN_PREF:
- end|ROUTE_LOAD_CALLEE_DOMAIN_PREF:
- start|ROUTE_FIND_CALLER:
- start|ROUTE_AUTH:
- exit|ROUTE_AUTH:
sip_in:
- '^INVITE sip:testuser1001@'
- 'Contact: sip:testuser1002@'
- 'To: <sip:testuser1001@'
- 'CSeq: 1 INVITE'
- 'Max-Forwards: 16'
- 'Content-Type: application/sdp'
sip_out:
- [
'^SIP/2.0 100 Trying',
'CSeq: 1 INVITE',
'From: <sip:testuser1002@',
'To: <sip:testuser1001@'
]
- [
'^SIP/2.0 407 Proxy Authentication Required',
'CSeq: 1 INVITE',
'From: <sip:testuser1002@',
'To: <sip:testuser1001@'
]

91
scenarios/invite_voicebox/0003_test.yml.tt2
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# INVITE_OFFLINE scenario
flow:
- start|MAIN:
- start|ROUTE_NET_INFO:
- end|ROUTE_NET_INFO:
- start|ROUTE_PRX_REQUEST:
- start|ROUTE_INVITE:
- start|ROUTE_LOAD_CALLEE_DOMAIN_PREF:
- end|ROUTE_LOAD_CALLEE_DOMAIN_PREF:
- start|ROUTE_FIND_CALLER:
- start|ROUTE_AUTH:
- start|ROUTE_ADD_CALLINFO_REPLY:
- end|ROUTE_ADD_CALLINFO_REPLY:
- end|ROUTE_AUTH:
- end|ROUTE_FIND_CALLER:
- start|ROUTE_LOAD_CALLER_PREF:
- start|ROUTE_CLEAR_CALLER_PREF:
- end|ROUTE_CLEAR_CALLER_PREF:
- end|ROUTE_LOAD_CALLER_PREF:
- start|ROUTE_GET_CALLER_CLI:
- end|ROUTE_GET_CALLER_CLI:
- start|ROUTE_GET_FORWARDER_CLI:
- end|ROUTE_GET_FORWARDER_CLI:
- start|ROUTE_CHECK_USERPROV_CLI:
- start|ROUTE_CHECK_CLI_ALLOWED:
- end|ROUTE_CHECK_CLI_ALLOWED:
- end|ROUTE_CHECK_USERPROV_CLI:
- start|ROUTE_CLIR:
- end|ROUTE_CLIR:
- start|ROUTE_FIND_CALLEE:
- start|ROUTE_CLEAR_CALLEE_PREF:
- end|ROUTE_CLEAR_CALLEE_PREF:
- start|ROUTE_NCOS:
- end|ROUTE_NCOS:
- start|ROUTE_NCOS:
- end|ROUTE_NCOS:
- start|ROUTE_BLOCK_OUT:
- end|ROUTE_BLOCK_OUT:
- start|ROUTE_LOAD_CALLEE_PREF:
- start|ROUTE_BLOCK_IN:
- end|ROUTE_BLOCK_IN:
- start|ROUTE_LOAD_CF_MAP:
- end|ROUTE_LOAD_CF_MAP:
- start|ROUTE_CHECK_CF_PERIOD:
- return|ROUTE_CHECK_CF_PERIOD:
- start|ROUTE_ACC_CF:
- start|ROUTE_ACC_CALLER:
- end|ROUTE_ACC_CALLER:
- start|ROUTE_ACC_CALLEE:
- end|ROUTE_ACC_CALLEE:
- end|ROUTE_ACC_CF:
- start|ROUTE_EXECUTE_CF_LOOP:
- start|BRANCH_ROUTE_FWD_LOOP:
- start|ROUTE_ADD_CALLINFO:
- end|ROUTE_ADD_CALLINFO:
- end|BRANCH_ROUTE_FWD_LOOP:
- exit|ROUTE_EXECUTE_CF_LOOP:
sip_in:
- '^INVITE'
- 'Contact: sip:testuser1002@'
- 'CSeq: 2 INVITE'
- 'Max-Forwards: 16'
- 'Content-Type: application/sdp'
- 'Proxy-Authorization: Digest username="testuser1002"'
sip_out:
- [
'^SIP/2.0 100 Trying',
'CSeq: 2 INVITE',
'From: <sip:testuser1002@',
'To: <sip:testuser1001@'
]
- [
'^SIP/2.0 101 Connecting',
'CSeq: 2 INVITE',
'From: <sip:testuser1002@',
'To: <sip:testuser1001@'
]
- [
'^INVITE sip:vmu4311001@voicebox.local SIP/2.0',
'CSeq: 2 INVITE',
'From: <sip:testuser1002@',
'To: <sip:testuser1001@',
'Content-Type: application/sdp',
'P-Caller-UUID: \w+',
'P-First-Caller-UPN: 4311002',
'P-First-Caller-NPN: 4311002',
'P-Acc-Caller-User: testuser1001',
'P-Acc-State: cfu',
'P-App-Param: ;profile=ngcp_cf',
'P-App-Name: sbc'
]

97
scenarios/invite_voicebox/0005_test.yml.tt2
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# INVITE_OFFLINE scenario
flow:
- start|MAIN:
- start|ROUTE_NET_INFO:
- end|ROUTE_NET_INFO:
- start|ROUTE_PRX_REQUEST:
- start|ROUTE_INVITE:
- start|ROUTE_LOAD_CALLEE_DOMAIN_PREF:
- end|ROUTE_LOAD_CALLEE_DOMAIN_PREF:
- start|ROUTE_FIND_CALLER:
- end|ROUTE_FIND_CALLER:
- start|ROUTE_LOAD_CALLER_PREF:
- start|ROUTE_CLEAR_CALLER_PREF:
- end|ROUTE_CLEAR_CALLER_PREF:
- end|ROUTE_LOAD_CALLER_PREF:
- start|ROUTE_GET_FORWARDER_CLI:
- end|ROUTE_GET_FORWARDER_CLI:
- start|ROUTE_CHECK_USERPROV_CLI:
- start|ROUTE_CHECK_CLI_ALLOWED:
- end|ROUTE_CHECK_CLI_ALLOWED:
- end|ROUTE_CHECK_USERPROV_CLI:
- start|ROUTE_CLIR:
- end|ROUTE_CLIR:
- start|ROUTE_FIND_CALLEE:
- start|ROUTE_VOICEBOX:
- start|ROUTE_OUTBOUND:
- start|ROUTE_ACC_CALLER:
- end|ROUTE_ACC_CALLER:
- start|ROUTE_ACC_CALLEE:
- end|ROUTE_ACC_CALLEE:
- start|BRANCH_ROUTE_NO_SBC:
- start|ROUTE_BRANCH_ACC_RTP:
- end|ROUTE_BRANCH_ACC_RTP:
- start|ROUTE_SET_CALLER_CLI:
- start|ROUTE_PREPARE_CALLER_CLI:
- end|ROUTE_PREPARE_CALLER_CLI:
- start|ROUTE_PREPARE_CALLER_CLI:
- end|ROUTE_PREPARE_CALLER_CLI:
- start|ROUTE_PREPARE_CALLER_CLI:
- end|ROUTE_PREPARE_CALLER_CLI:
- start|ROUTE_PREPARE_CALLER_CLI:
- end|ROUTE_PREPARE_CALLER_CLI:
- start|ROUTE_PREPARE_CALLER_CLI:
- end|ROUTE_PREPARE_CALLER_CLI:
- end|ROUTE_SET_CALLER_CLI:
- start|ROUTE_SET_CALLEE:
- end|ROUTE_SET_CALLEE:
- start|ROUTE_ADD_CALLINFO:
- end|ROUTE_ADD_CALLINFO:
- end|BRANCH_ROUTE_NO_SBC:
- exit|ROUTE_OUTBOUND:
- start|REPLY_ROUTE_NAT:
- exit|REPLY_ROUTE_NAT:
- start|REPLY_ROUTE_NAT:
- exit|REPLY_ROUTE_NAT:
sip_in:
- '^INVITE sip:vmu4311001@voicebox.local SIP/2.0'
- 'Contact: <sip:127.0.0.1:5080>'
- 'CSeq: 10 INVITE'
- 'Content-Type: application/sdp'
- 'P-Caller-UUID: \w+'
- 'P-First-Caller-UPN: 4311002'
- 'P-First-Caller-NPN: 4311002'
- 'P-Acc-Caller-User: testuser1001'
- 'P-Acc-State: cfu'
sip_out:
- [
'^SIP/2.0 100 Trying',
'CSeq: 10 INVITE',
'From: <sip:testuser1002@',
'To: <sip:testuser1001@'
]
- [
'^INVITE sip:vmu4311001@voicebox.local SIP/2.0',
'CSeq: 10 INVITE',
'From: <sip:4311002@',
'To: <sip:testuser1001@',
'Contact: <sip:127.0.0.1:5080>',
'Content-Type: application/sdp'
]
- [
'^SIP/2.0 200 OK',
'CSeq: 10 INVITE',
'From: <sip:testuser1002@',
'To: <sip:testuser1001@',
'Contact: <sip:vmu4311001@127.0.0.1:5070>',
'User-Agent: NGCP Voicebox Server',
'Content-Type: application/sdp'
]
- [
'^SIP/2.0 200 OK',
'CSeq: 2 INVITE',
'From: <sip:testuser1002@',
'To: <sip:testuser1001@',
'Contact: <sip:vmu4311001@127.0.0.1:5080>',
'Content-Type: application/sdp'
]

25
scenarios/invite_voicebox/0009_test.yml.tt2
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# INVITE_OFFLINE scenario
flow:
- start|MAIN:
- start|ROUTE_NET_INFO:
- end|ROUTE_NET_INFO:
- start|ROUTE_PRX_REQUEST:
- start|ROUTE_OUTBOUND:
sip_in:
- '^ACK sip:127.0.0.1:5080 SIP/2.0'
- 'Contact: sip:testuser1002@'
- 'CSeq: 2 ACK'
- 'Max-Forwards: 16'
- 'Content-Length: 0'
- 'From: <sip:testuser1002@'
- 'To: <sip:testuser1001@'
sip_out:
- [
'^ACK sip:127.0.0.1:5080 SIP/2.0',
'Contact: sip:testuser1002@',
'CSeq: 2 ACK',
'Max-Forwards: 15',
'Content-Length: 0',
'From: <sip:testuser1002@',
'To: <sip:testuser1001@'
]

24
scenarios/invite_voicebox/0010_test.yml.tt2
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# INVITE_OFFLINE scenario
flow:
- start|MAIN:
- start|ROUTE_NET_INFO:
- end|ROUTE_NET_INFO:
- start|ROUTE_PRX_REQUEST:
- start|ROUTE_OUTBOUND:
sip_in:
- '^ACK sip:vmu4311001@127.0.0.1:5070 SIP/2.0'
- 'Contact: <sip:127.0.0.1:5080>'
- 'CSeq: 10 ACK'
- 'Content-Length: 0'
- 'From: <sip:testuser1002@'
- 'To: <sip:testuser1001@'
sip_out:
- [
'^ACK sip:vmu4311001@127.0.0.1:5070 SIP/2.0',
'Contact: <sip:127.0.0.1:5080>',
'CSeq: 10 ACK',
'Max-Forwards: 10',
'Content-Length: 0',
'From: <sip:4311002@',
'To: <sip:testuser1001@'
]

46
scenarios/invite_voicebox/0011_test.yml.tt2
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# INVITE_OFFLINE scenario
flow:
- start|MAIN:
- start|ROUTE_NET_INFO:
- end|ROUTE_NET_INFO:
- start|ROUTE_PRX_REQUEST:
- start|ROUTE_STOP_RTPPROXY:
- end|ROUTE_STOP_RTPPROXY:
- start|ROUTE_OUTBOUND:
- exit|ROUTE_OUTBOUND:
# 200 OK
- start|REPLY_ROUTE_NAT:
- exit|REPLY_ROUTE_NAT:
sip_in:
- '^BYE sip:127.0.0.1:5080 SIP/2.0'
- 'Contact: sip:testuser1002@'
- 'CSeq: 3 BYE'
- 'Max-Forwards: 16'
- 'Content-Length: 0'
- 'From: <sip:testuser1002@'
- 'To: <sip:testuser1001@'
sip_out:
- [
'^SIP/2.0 100 Trying',
'CSeq: 3 BYE',
'Content-Length: 0',
'From: <sip:testuser1002@',
'To: <sip:testuser1001@'
]
- [
'^BYE sip:127.0.0.1:5080 SIP/2.0',
'Contact: sip:testuser1002@',
'CSeq: 3 BYE',
'Max-Forwards: 15',
'Content-Length: 0',
'From: <sip:testuser1002@',
'To: <sip:testuser1001@'
]
- [
'^SIP/2.0 200 OK',
'CSeq: 3 BYE',
'Content-Length: 0',
'From: <sip:testuser1002@',
'To: <sip:testuser1001@'
]

44
scenarios/invite_voicebox/0013_test.yml.tt2
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# INVITE_OFFLINE scenario
flow:
- start|MAIN:
- start|ROUTE_NET_INFO:
- end|ROUTE_NET_INFO:
- start|ROUTE_PRX_REQUEST:
- start|ROUTE_STOP_RTPPROXY:
- end|ROUTE_STOP_RTPPROXY:
- start|ROUTE_OUTBOUND:
- exit|ROUTE_OUTBOUND:
# 200 OK
- start|REPLY_ROUTE_NAT:
- exit|REPLY_ROUTE_NAT:
sip_in:
- '^BYE sip:vmu4311001@127.0.0.1:5070 SIP/2.0'
- 'CSeq: 11 BYE'
- 'Content-Length: 0'
- 'From: <sip:testuser1002@'
- 'To: <sip:testuser1001@'
sip_out:
- [
'^SIP/2.0 100 Trying',
'CSeq: 11 BYE',
'Content-Length: 0',
'From: <sip:testuser1002@',
'To: <sip:testuser1001@'
]
- [
'^BYE sip:vmu4311001@127.0.0.1:5070 SIP/2.0',
'CSeq: 11 BYE',
'Max-Forwards: 10',
'Content-Length: 0',
'From: <sip:4311002@',
'To: <sip:testuser1001@'
]
- [
'^SIP/2.0 200 OK',
'CSeq: 11 BYE',
'Content-Length: 0',
'From: <sip:testuser1002@',
'To: <sip:testuser1001@',
'User-Agent: NGCP Voicebox Server'
]

16
scenarios/invite_voicebox/callforward.yml
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---
testuser1001@spce.test:
destination_set:
- destinations:
- destination: sip:vmu4311001@voicebox.local
priority: 0
timeout: 2
name: cf_set_1001_00
map:
- destination_set_id: cf_set_1001_00
type: cfu
voicebox:
password: 0000
email: root@spce.test
attach: 0
delete: 1

22
scenarios/invite_voicebox/scenario.yml
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subscribers:
spce.test:
testuser1001:
password: testuser
cc: 43
ac: 1
sn: 1001
testuser1002:
password: testuser
cc: 43
ac: 1
sn: 1002
scenarios:
- ip: 127.126.0.1
username: testuser1002
domain: spce.test
responders:
- ip: 127.1.0.1
username: testuser1001
domain: spce.test
register: no
active: no

146
scenarios/invite_voicebox/sipp_scenario00.xml
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<?xml version="1.0" encoding="ISO-8859-1" ?>
<scenario name="Sipwise NGCP Benchmark UAC Caller">
<send start_rtd="1" start_rtd="2">
<![CDATA[
INVITE sip:[field0 file="callee.csv" line=0]@[field3 file="callee.csv" line=0] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: <sip:[field0 file="caller.csv"]@[field2 file="caller.csv"]>;tag=[pid]SIPpTag00[call_number]
To: <sip:[field0 file="callee.csv" line=0]@[field3 file="callee.csv" line=0]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:[field0 file="caller.csv"]@[local_ip]:[local_port]
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:50
]]>
</send>
<recv response="100" rtd="1" optional="true">
</recv>
<recv response="407" rtd="2" auth="true"/>
<send>
<![CDATA[
ACK sip:[field0 file="callee.csv" line=0]@[field3 file="callee.csv" line=0]:[remote_port] SIP/2.0
[last_Via:]
From: <sip:[field0 file="caller.csv"]@[field2 file="caller.csv"]>;tag=[pid]SIPpTag00[call_number]
To: <sip:[field0 file="callee.csv" line=0]@[field3 file="callee.csv" line=0]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:[field0 file="caller.csv"]@[local_ip]:[local_port]
Max-Forwards: 70
Content-Length: 0
]]>
</send>
<pause milliseconds="500"/>
<send start_rtd="3">
<![CDATA[
INVITE sip:[field0 file="callee.csv" line=0]@[field3 file="callee.csv" line=0] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: <sip:[field0 file="caller.csv"]@[field2 file="caller.csv"]>;tag=[pid]SIPpTag00[call_number]
To: <sip:[field0 file="callee.csv" line=0]@[field3 file="callee.csv" line=0]>
Call-ID: [call_id]
CSeq: 2 INVITE
Contact: sip:[field0 file="caller.csv"]@[local_ip]:[local_port]
Max-Forwards: 70
[field1 file="caller.csv"]
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:50
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="180" optional="true">
</recv>
<recv response="183" optional="true">
</recv>
<recv response="200" rrs="true" rtd="3">
</recv>
<send>
<![CDATA[
ACK [next_url] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: <sip:[field0 file="caller.csv"]@[field2 file="caller.csv"]>;tag=[pid]SIPpTag00[call_number]
To: <sip:[field0 file="callee.csv" line=0]@[field3 file="callee.csv" line=0]>[peer_tag_param]
Call-ID: [call_id]
[routes]
CSeq: 2 ACK
Contact: sip:[field0 file="caller.csv"]@[local_ip]:[local_port]
Max-Forwards: 70
Content-Length: 0
]]>
</send>
<pause milliseconds="500"/>
<nop>
<action>
<exec play_pcap_audio="scenarios/media181sec.pcap"/>
</action>
</nop>
<pause milliseconds="500"/>
<send start_rtd="4">
<![CDATA[
BYE [next_url] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: <sip:[field0 file="caller.csv"]@[field2 file="caller.csv"]>;tag=[pid]SIPpTag00[call_number]
To: <sip:[field0 file="callee.csv" line=0]@[field3 file="callee.csv" line=0]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 3 BYE
Contact: sip:[field0 file="caller.csv"]@[local_ip]:[local_port]
[routes]
Max-Forwards: 70
Content-Length: 0
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="200" crlf="true" rtd="4">
</recv>
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200, 500, 1000"/>
<!-- <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/> >
</scenario>
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