Transcript VoIP SIP

United Networking
4.16 VoIP SIP Configuration Examples
v1.0 2008
www.oneaccess-net.com
[email protected]
13 April, 2015
Copyright © OneAccess Networks – All rights reserved
Configuration examples
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

Gateway to gateway call (uac/uas, no SIP server).
Register & Invite methods over proxy server:



40x challenge for Register & Invite methods :



Register transaction: Slides 62 & 63
Invite transaction: Slides 64 to 65
Authenticated register transaction: Slides 68 to 70
Authenticated Invite transaction: Slides 71 to 77
Voice backup
2
Gateway-to-Gateway Call (1)
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 Gateway
to Gateway call (1)
.2
#200
Up
#100
.1
Down
192.168.1.0/24
3
Gateway-to-Gateway Call (2)
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hostname UP
interface FastEthernet 0/0
ip address 192.168.1.1 255.255.255.0
exit
interface bri 5/0
isdn
protocol-emulation isdn-nt
exit
no shutdown
execute
exit
snmp set-write-community private
snmp set-read-community public
voice-default
voice-port 5/0
clock-source free_run
exit
dial-peer voice pots 0
pots-group 0
port 5/0
no shutdown
exit
dial-peer voice voip 0
sig-protocol sip
gw-ip-address 192.168.1.2
voip-coder-profile 0
no shutdown
exit
voice-routing
route 1
dial-peer pots-group 0
prefix 100 length 3
prefix-type outgoing called last
exit
route 2
dial-peer voip 0
prefix 200 length 3
prefix-type outgoing called last
exit
exit
sip-gateway
gw-interface fastethernet 0 intrusive
no shutdown
exit
voip-coder-profile 0
codec 0 g711a 20
exit
4
Gateway-to-Gateway Call (3)
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hostname DOWN
interface FastEthernet 0/0
ip address 192.168.1.2 255.255.255.0
exit
interface bri 5/0
isdn
protocol-emulation isdn-nt
exit
no shutdown
execute
exit
snmp set-write-community private
snmp set-read-community public
voice-default
voice-port 5/0
clock-source free_run
exit
dial-peer voice pots 0
pots-group 0
port 5/0
no shutdown
exit
dial-peer voice voip 0
sig-protocol sip
gw-ip-address 192.168.1.1
voip-coder-profile 0
no shutdown
exit
voice-routing
route 1
dial-peer pots-group 0
prefix 200 length 3
prefix-type outgoing called last
exit
route 2
dial-peer voip 0
prefix 100 length 3
prefix-type outgoing called last
exit
exit
sip-gateway
gw-interface fastethernet 0/0 intrusive
no shutdown
exit
voip-coder-profile 0
codec 0 g711a 20
exit
5
Register and Invite methods over Proxy SIP (1)
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Proxy/registrar server
@oneaccess-net.com
.69.2
172.17.68.0/22
.69.1
REMOTE_SITE
.100
MAIN_SITE
.2
PABX BRI 5/0
IP network
.100
.10
192.0.0.0/24
BRI 5/0
BRI 5/1
Pabx
192.100.100.0/24
#300
#0141877446
#301
6
Register and Invite methods over Proxy SIP (2)
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hostname MAIN_SITE
Ip route 0.0.0.0 0.0.0.0 192.100.100.100
Ip host oneaccess-net.com
interface FastEthernet 0/0
ip address 192.100.100.2 255.255.255.0
exit
interface bri 5/0
isdn
protocol-emulation isdn-nt
tei-negotiation static
exit
no shutdown
execute
exit
voice-default
voice-port 5/0
clock-source free_run
exit
dial-peer voice pots 0
pots-group 0
port 5/0
no shutdown
exit
dial-peer voice voip 0
sig-protocol sip
voip-coder-profile 0
no shutdown
exit
voice-routing
route 1
dial-peer pots-group 0 ua-sip
prefix 300 length 3
prefix-type outgoing called last
exit
route 2
dial-peer voip 0
prefix 301 length 3
prefix-type outgoing called last
exit
exit
sip-gateway
outbound-proxy 172.17.69.2
reg-dns-add oneaccess-net.com
prox-dns-add oneaccess-net.com
gw-interface fastethernet 0/0 intrusive
device-host-name oneaccess-net.com
uri-contact ip-address
no shutdown
exit
voip-coder-profile 0
codec 0 g711a 20
7
Register and Invite methods over Proxy SIP (3)
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hostname REMOTE_SITE
Ip route 0.0.0.0 0.0.0.0 192.100.100.100
interface FastEthernet 0/0
ip address 192.0.0.10 255.255.255.0
exit
interface bri 5/0
isdn
protocol-emulation isdn-nt
tei-negotiation static
exit
no shutdown
execute
exit
voice-default
voice-port 5/0
clock-source free_run
exit
dial-peer voice pots 0
pots-group 0
port 5/0
inser-calling-number 301
no shutdown
exit
dial-peer voice voip 0
sig-protocol sip
voip-coder-profile 0
no shutdown
exit
voice-routing
route 1
dial-peer pots-group 0 ua-sip
prefix 301 length 3
prefix-type outgoing called last
exit
route 2
dial-peer voip 0
prefix 300 length 3
prefix-type outgoing called last
exit
exit
sip-gateway
outbound-proxy 172.17.69.2
reg-dns-add oneaccess-net.com
prox-dns-add oneaccess-net.com
gw-interface fastethernet 0/0 intrusive
device-host-name oneaccess-net.com
uri-contact ip-address
no shutdown
exit
voip-coder-profile 0
codec 0 g711a 20
exit
8
SIP Registrar server (4)
debug sip traces about Register method
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1- Internet Protocol, Src Addr: 192.100.100.2 (192.100.100.2), Dst Addr: 172.17.69.2 (172.17.69.2)
Request-Line: REGISTER sip:oneaccess-net.com:5060 SIP/2.0
Method: REGISTER
Resent Packet: False
Message Header
Call-ID: 31
Contact: <sip:[email protected]:5060;user=phone>
Content-Type: application/*
CSeq: 56 REGISTER
Expires: 1800
From: <sip:[email protected]:5060;user=phone>;tag=22C8
Max-Forwards: 70
Privacy: none
To: <sip:[email protected];user=phone>
Via: SIP/2.0/UDP 192.100.100.2:5060;branch=z9hG4bK-556D-4F
Content-Length: 0
2- Internet Protocol, Src Addr: 172.17.69.2 (172.17.69.2), Dst Addr: 192.100.100.2 (192.100.100.2)
Status-Line: SIP/2.0 100 Trying
Status-Code: 100
Resent Packet: False
Message Header
Via: SIP/2.0/UDP 192.100.100.2:5060;branch=z9hG4bK-556D-4F
From: <sip:[email protected]:5060;user=phone>;tag=22C8
To: <sip:[email protected];user=phone>
Call-ID: 31
CSeq: 56 REGISTER
Server: Brekeke OnDO SIP Server (1.2.4.3/144)
9
SIP Registrar server (5)
debug sip traces about Register method
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Frame 3
Internet Protocol, Src Addr: 172.17.69.2 (172.17.69.2), Dst Addr: 192.100.100.2 (192.100.100.2)
Session Initiation Protocol
Status-Line: SIP/2.0 200 OK
Status-Code: 200
Resent Packet: False
Message Header
Via: SIP/2.0/UDP 192.100.100.2:5060;branch=z9hG4bK-556D-4F
From: <sip:[email protected]:5060;user=phone>;tag=22C8
To: <sip:[email protected];user=phone>;tag=1136463813166-493603485
Call-ID: 31
CSeq: 56 REGISTER
Contact: <sip:[email protected]:5060>;expires=1800;q=1.0
Server: Brekeke OnDO SIP Server (1.2.4.3/144)
Content-Length: 0
Another SIP Register Method (new Call-ID & new Cseq) is generated
about the « user » 0141877456. Three same frames as for « 300 »
are sent to oneaccess-net.com Registrar server.
10
SIP Proxy server (6)
debug sip trace about Invite method
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Session Initiation Protocol
Request-Line: INVITE sip:[email protected]:5060;user=phone SIP/2.0
Method: INVITE
Resent Packet: False
Message Header
Call-ID: 3C
Contact: <sip:[email protected]:5060;user=phone>
Content-Type: application/sdp
CSeq: 66 INVITE
From: <sip:[email protected]:5060;user=phone>;tag=5548
Privacy: none
Supported: replaces
To: <sip:[email protected];user=phone>
Via: SIP/2.0/UDP 192.100.100.2:5060;branch=z9hG4bK-3418-59
Content-Length: 150
Message body
Session Description Protocol
11
Transactions about an Invite method (7)
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No.
Source
1
192.100.100.2
Destination
172.17.69.2
Protocol
SIP/SDP
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
192.100.100.2
192.0.0.10
172.17.69.2
192.100.100.2
172.17.69.2
192.100.100.2
172.17.69.2
192.100.100.2
172.17.69.2
192.0.0.10
192.100.100.2
192.0.0.10
172.17.69.2
192.0.0.10
172.17.69.2
192.100.100.2
SIP
SIP/SDP
SIP
SIP
SIP
SIP
SIP/SDP
SIP/SDP
SIP
SIP
RTP
RTP
SIP
SIP
SIP
SIP
172.17.69.2
172.17.69.2
192.0.0.10
172.17.69.2
192.0.0.10
172.17.69.2
192.0.0.10
172.17.69.2
192.100.100.2
172.17.69.2
192.0.0.10
192.100.100.2
192.100.100.2
172.17.69.2
192.0.0.10
172.17.69.2
Info
Request: INVITE sip:[email protected]:5060;
user=phone, with sdp
Status: 100 Trying
Request: INVITE sip:[email protected]:5060, with sdp
Status: 183 Proceeding
Status: 183 Proceeding
Status: 180 Ringing
Status: 180 Ringing
Status: 200 OK, with session description
Status: 200 OK, with session description
Request: ACK sip:[email protected]:5060;user=phone
Request: ACK sip:[email protected]:5060
Payload type=ITU-T G.711 PCMA, SSRC=79034265, Seq=1554
Payload type=ITU-T G.711 PCMA, SSRC=190383165, Seq=1025
Request: BYE sip:[email protected]:5060;user=phone
Request: BYE sip:[email protected]:5060
Status: 200 OK
Status: 200 OK
12
Register and Invite methods including 40x challenge (1)
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hostname MAIN_SITE
sip-gateway
Ip host oneaccess-net.com
outbound-proxy 172.17.69.2
dial-peer voice voip 0
reg-dns-add oneaccess-net.com
sig-protocol sip
prox-dns-add oneaccess-net.com
voip-coder-profile 0
gw-interface fastethernet 0/0 intrusive
no shutdown
device-host-name oneaccess-net.com
exit
uri-contact ip-address
voice-routing
no shutdown
route 1
exit
dial-peer pots-group 0 ua-sip
sip-authentication isdn_call 300
Digest username and password for
prefix 300 length 3
Authentication purpose at Register and
prefix-type outgoing called last Invite Methods
exit
route 2
dial-peer pots-group 1 ua-sip
Setting of route 2 doesn’t support
prefix 0141877446 length 10
authentication request
prefix-type outgoing called last
exit
route 3
dial-peer voip 0
prefix 301 length 3
prefix-type outgoing called last
exit
exit
13
Register and Invite methods including 40x challenge (2)
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hostname REMOTE_SITE
Ip host oneaccess-net.com
dial-peer voice voip 0
sig-protocol sip
voip-coder-profile 0
no shutdown
exit
voice-routing
route 1
dial-peer pots-group 0 ua-sip
sip-authentication remote301 301
prefix 301 length 3
prefix-type outgoing called last
exit
route 2
dial-peer voip 0
prefix 300 length 3
prefix-type outgoing called last
exit
exit
sip-gateway
outbound-proxy 172.17.69.2
reg-dns-add oneaccess-net.com
prox-dns-add oneaccess-net.com
gw-interface fastethernet 0/0 intrusive
device-host-name oneaccess-net.com
uri-contact ip-address
no shutdown
exit
Digest username and password for
Authentication purpose about Register
and Invite methods.
14
Register method including 401 challenge (3)
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1-Src Addr: 192.100.100.2 (192.100.100.2), Dst Addr: 172.17.69.2 (172.17.69.2)
Request-Line: REGISTER sip:oneaccess-net.com:5060 SIP/2.0
Message Header
Call-ID: 4
Contact: <sip:[email protected]:5060;user=phone>
CSeq: 13 REGISTER
Expires: 1800
From: <sip:[email protected]:5060;user=phone>;tag=3E49
To: <sip:[email protected];user=phone>
Via: SIP/2.0/UDP 192.100.100.2:5060;branch=z9hG4bK-70C8-D
2-Src Addr: 192.100.100.2 (192.100.100.2), Dst Addr: 172.17.69.2 (172.17.69.2)
Request-Line: REGISTER sip:oneaccess-net.com:5060 SIP/2.0
Message Header
Call-ID: 5
Contact: <sip:[email protected]:5060;user=phone>
CSeq: 14 REGISTER
Expires: 1800
From: <sip:0141877446@ oneaccess-net.com :5060;user=phone>;tag=6E81
To: <sip:[email protected];user=phone>
Via: SIP/2.0/UDP 192.100.100.2:5060;branch=z9hG4bK-3E5D-E
Content-Length: 0
3-Src Addr: 172.17.69.2 (172.17.69.2), Dst Addr: 192.100.100.2 (192.100.100.2)
Status-Line: SIP/2.0 100 Trying
Message Header
To: <sip:[email protected];user=phone>
Call-ID: 4
CSeq: 13 REGISTER
15
Register method including 401 challenge (4)
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4-Src Addr: 172.17.69.2 (172.17.69.2), Dst Addr: 192.100.100.2 (192.100.100.2)
Status-Line: SIP/2.0 100 Trying
Message Header
To: <sip:0141877446@ oneaccess-net.com;user=phone>
Call-ID: 5
CSeq: 14 REGISTER
5-Src Addr: 172.17.69.2 (172.17.69.2), Dst Addr: 192.100.100.2 (192.100.100.2)
Status-Line: SIP/2.0 401 Unauthorized
Message Header
To: <sip:300@ oneaccess-net.com;user=phone>;tag=1136480194441-490786848
Call-ID: 4
CSeq: 13 REGISTER
WWW-Authenticate: Digest realm="LLA009-FFG890J",nonce="4f9bcecd34f542674791d2e7fcc1dd5f70bf6265"
6-Src Addr: 192.100.100.2 (192.100.100.2), Dst Addr: 172.17.69.2 (172.17.69.2)
Request-Line: REGISTER sip:oneaccess-net.com:5060 SIP/2.0
Message Header
Authorization: Digest username="isdn_call",realm="LLA009-FFG890J",nonce="4f9bcecd34f542674791d2e7fcc1dd5f70bf6265",
uri="oneaccess-net.com",response="df0963cf7aa95ad03719095738c65c56",nc=00000000
Call-ID: 4
CSeq: 15 REGISTER
From: <sip:300@ oneaccess-net.com :5060;user=phone>;tag=4EC5
7-Src Addr: 192.100.100.2 (192.100.100.2), Dst Addr: 172.17.69.2 (172.17.69.2)
Request-Line: REGISTER sip:oneaccess-net.com:5060 SIP/2.0
Message Header
Authorization: Digest username="0141877446",realm="LLA009FFG890J",nonce="45a6856be7e33dcf40079dd59ba0fb7370bf6265",
uri="oneaccess-net.com",response="db73863d0297b3e827cfdbfe9561eb2d",nc=00000000
Call-ID: 5
CSeq: 16 REGISTER
From: <sip:0141877446@ oneaccess-net.com :5060;user=phone>;tag=2EE5
16
Register method including 401 challenge (5)
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8-Src Addr: 172.17.69.2 (172.17.69.2), Dst Addr: 192.100.100.2 (192.100.100.2)
Status-Line: SIP/2.0 200 OK
Message Header
To: <sip:[email protected];user=phone>;tag=1136480195282-34834762
Call-ID: 4
CSeq: 15 REGISTER
9-Src Addr: 172.17.69.2 (172.17.69.2), Dst Addr: 192.100.100.2 (192.100.100.2)
Status-Line: SIP/2.0 403 Forbidden
Message Header
To: <sip:[email protected];user=phone>;tag=1136480195292-792478172
Call-ID: 5
CSeq: 16 REGISTER
17
Invite method including 407 challenge (6)
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1- Src Addr: 192.100.100.2 (192.100.100.2), Dst Addr: 172.17.69.2 (172.17.69.2)
Request-Line: INVITE sip:[email protected]:5060;user=phone SIP/2.0
Message Header
Call-ID: 10
Contact: <sip:[email protected]:5060;user=phone>
CSeq: 47 INVITE
From: <sip:[email protected] :5060;user=phone>;tag=23E
To: <sip:[email protected];user=phone>
Via: SIP/2.0/UDP 192.100.100.2:5060;branch=z9hG4bK-67B-33
Message body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): UserA 2890844526 2890844526 IN IP4 192.100.100.2
Session Name (s): Session SDP
Connection Information (c): IN IP4 192.100.100.2
Time Description, active time (t): 0 0
Media Description, name and address (m): audio 16390 RTP/AVP 8
Media Attribute (a): rtpmap:8 PCMA/8000
2- Src Addr: 172.17.69.2 (172.17.69.2), Dst Addr: 192.100.100.2 (192.100.100.2)
Status-Line: SIP/2.0 100 Trying
Message Header
Via: SIP/2.0/UDP 192.100.100.2:5060;branch=z9hG4bK-67B-33
From: <sip:[email protected] :5060;user=phone>;tag=23E
To: <sip:[email protected];user=phone>
Call-ID: 10
CSeq: 47 INVITE
18
Invite method including 407 challenge (8)
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3- Src Addr: 172.17.69.2 (172.17.69.2), Dst Addr: 192.100.100.2 (192.100.100.2)
Status-Line: SIP/2.0 407 Proxy Authentication Required
Message Header
Via: SIP/2.0/UDP 192.100.100.2:5060;branch=z9hG4bK-67B-33
From: <sip:[email protected] :5060;user=phone>;tag=23E
To: <sip:[email protected];user=phone>;tag=1136483382615-833403183
Call-ID: 10
CSeq: 47 INVITE
Server: Brekeke OnDO SIP Server (1.2.4.3/144)
Proxy-Authenticate: Digest realm="LLA009-FFG890J",nonce="2081e4ec87b4bfe157c3466950e08a4f70bf6265«
4- Src Addr: 192.100.100.2 (192.100.100.2), Dst Addr: 172.17.69.2 (172.17.69.2)
Request-Line: ACK sip:[email protected]:5060;user=phone SIP/2.0
Message Header
Call-ID: 10
CSeq: 47 ACK
From: <sip:[email protected] :5060;user=phone>;tag=23E
To: <sip:[email protected];user=phone>;tag=1136483382615-833403183
Via: SIP/2.0/UDP 192.100.100.2:5060;branch=z9hG4bK-67B-33
5- Src Addr: 192.100.100.2 (192.100.100.2), Dst Addr: 172.17.69.2 (172.17.69.2)
Request-Line: INVITE sip:[email protected]:5060;user=phone SIP/2.0
Message Header
Call-ID: 10
Contact: <sip:[email protected]:5060;user=phone>
CSeq: 48 INVITE
From: <sip:[email protected] :5060;user=phone>;tag=23E
Proxy-Authorization: Digest username="isdn_call2",realm="LLA009-FFG890J",nonce="2081e4ec87b4bfe157c3466950e08a4f70bf6265",
uri="oneaccess-net.com",response="e6f6da842a1ab64e06dc1bd621fa5fb1",nc=00000000
To: <sip:[email protected];user=phone>
Via: SIP/2.0/UDP 192.100.100.2:5060;branch=z9hG4bK-49F9-34
Content-Length: 150
Message body : Session Description Protocol
19
Invite method including 407 challenge (9)
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6- Src Addr: 172.17.69.2 (172.17.69.2), Dst Addr: 192.100.100.2 (192.100.100.2)
Status-Line: SIP/2.0 100 Trying
Message Header
Via: SIP/2.0/UDP 192.100.100.2:5060;branch=z9hG4bK-49F9-34
From: <sip:[email protected] :5060;user=phone>;tag=23E
To: <sip:[email protected];user=phone>
Call-ID: 10
CSeq: 48 INVITE
7- Src Addr: 172.17.69.2 (172.17.69.2), Dst Addr: 192.0.0.10 (192.0.0.10)
Request-Line: INVITE sip:[email protected]:5060 SIP/2.0
Message Header
Call-ID: 10
Contact: <sip:[email protected] :5060;user=phone>
CSeq: 48 INVITE
From: <sip:[email protected];user=phone>;tag=23E
Proxy-Authorization: Digest username="isdn_call2",realm="LLA009-FFG890J",nonce="2081e4ec87b4bfe157c3466950e08a4f70bf6265",
uri="oneaccess-net.com",response="e6f6da842a1ab64e06dc1bd621fa5fb1",nc=00000000
To: <sip:[email protected] :5060;user=phone>
Via: SIP/2.0/UDP 172.17.69.2:5060;branch=z9hG4bKd2ca899425124.1
Via: SIP/2.0/UDP 192.100.100.2:5060;branch=z9hG4bK-49F9-34
Record-Route: <sip:172.17.69.2:5060;lr>
Content-Length: 150
Message body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): UserA 2890844526 2890844526 IN IP4 192.100.100.2
Session Name (s): Session SDP
Connection Information (c): IN IP4 192.100.100.2
Time Description, active time (t): 0 0
Media Description, name and address (m): audio 16390 RTP/AVP 8
Media Attribute (a): rtpmap:8 PCMA/8000
20
Invite method including 407 challenge (10)
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8- Src Addr: 192.0.0.10 (192.0.0.10), Dst Addr: 172.17.69.2 (172.17.69.2)
Status-Line: SIP/2.0 100 trying
Message Header
Call-ID: 10
CSeq: 48 INVITE
From: <sip:[email protected];user=phone>;tag=23E
Record-Route: <sip:172.17.69.2:5060;lr>
To: <sip:[email protected] :5060;user=phone>
Via: SIP/2.0/UDP 172.17.69.2:5060;received=172.17.69.2;branch=z9hG4bKd2ca899425124.1,SIP/2.0/UDP 192.100.100.2:5060;branch=z9hG4bK-49F9-34
9- Src Addr: 192.0.0.10 (192.0.0.10), Dst Addr: 172.17.69.2 (172.17.69.2)
Status-Line: SIP/2.0 183 Proceeding
Message Header
Call-ID: 10
CSeq: 48 INVITE
From: <sip:[email protected];user=phone>;tag=23E
Record-Route: <sip:172.17.69.2:5060;lr>
To: <sip:[email protected] :5060;user=phone>;tag=1469
Via: SIP/2.0/UDP 172.17.69.2:5060;received=172.17.69.2;branch=z9hG4bKd2ca899425124.1,SIP/2.0/UDP 192.100.100.2:5060;branch=z9hG4bK-49F9-34
10- Src Addr: 172.17.69.2 (172.17.69.2), Dst Addr: 192.100.100.2 (192.100.100.2)
Status-Line: SIP/2.0 183 Proceeding
Message Header
Call-ID: 10
CSeq: 48 INVITE
From: <sip:[email protected] :5060;user=phone>;tag=23E
Record-Route: <sip:172.17.69.2:5060;lr>
To: <sip:[email protected];user=phone>;tag=1469
Via: SIP/2.0/UDP 192.100.100.2:5060;branch=z9hG4bK-49F9-34
21
Invite method including 407 challenge (11)
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11- Src Addr: 192.0.0.10 (192.0.0.10), Dst Addr: 172.17.69.2 (172.17.69.2)
Status-Line: SIP/2.0 180 Ringing
Message Header
Call-ID: 10
CSeq: 48 INVITE
From: <sip:[email protected];user=phone>;tag=23E
Record-Route: <sip:172.17.69.2:5060;lr>
To: <sip:[email protected] :5060;user=phone>;tag=1469
Via: SIP/2.0/UDP172.17.69.2:5060;received=172.17.69.2;branch=z9hG4bKd2ca899425124.1,SIP/2.0/UDP;
192.100.100.2:5060;branch=z9hG4bK49F9
12- Src Addr: 172.17.69.2 (172.17.69.2), Dst Addr: 192.100.100.2 (192.100.100.2)
Status-Line: SIP/2.0 180 Ringing
Message Header
Call-ID: 10
CSeq: 48 INVITE
From: <sip:[email protected] :5060;user=phone>;tag=23E
Record-Route: <sip:172.17.69.2:5060;lr>
To: <sip:[email protected];user=phone>;tag=1469
Via: SIP/2.0/UDP 192.100.100.2:5060;branch=z9hG4bK-49F9-34
22
Invite method including 401 challenge (12)
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13- Src Addr: 192.0.0.10 (192.0.0.10), Dst Addr: 172.17.69.2 (172.17.69.2)
Status-Line: SIP/2.0 200 OK
Message Header
Call-ID: 10
Contact: <sip:[email protected]:5060;user=phone>
Content-Type: application/sdp
CSeq: 48 INVITE
From: <sip:[email protected];user=phone>;tag=23E
Record-Route: <sip:172.17.69.2:5060;lr>
To: <sip:[email protected] :5060;user=phone>;tag=1469
Via: SIP/2.0/UDP 172.17.69.2:5060;received=172.17.69.2;branch=z9hG4bKd2ca899425124.1,SIP/2.0/UDP 192.100.100.2:5060;branch=z9hG4bK-49
Message body: Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): UserA 2890844526 2890844526 IN IP4 192.0.0.10
Session Name (s): Session SDP
Connection Information (c): IN IP4 192.0.0.10
Time Description, active time (t): 0 0
Media Description, name and address (m): audio 16430 RTP/AVP 8
Media Attribute (a): rtpmap:8 PCMA/8000
14- Src Addr: 172.17.69.2 (172.17.69.2), Dst Addr: 192.100.100.2 (192.100.100.2)
Status-Line: SIP/2.0 200 OK
Message Header
Call-ID: 10
Contact: <sip:[email protected]:5060;user=phone>
CSeq: 48 INVITE
From: <sip:[email protected] :5060;user=phone>;tag=23E
Record-Route: <sip:172.17.69.2:5060;lr>
To: <sip:[email protected];user=phone>;tag=1469
Via: SIP/2.0/UDP 192.100.100.2:5060;branch=z9hG4bK-49F9-34
Message body: Session Description Protocol
23
Invite method including 401 challenge (13)
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15- Src Addr: 192.100.100.2 (192.100.100.2), Dst Addr: 172.17.69.2 (172.17.69.2)
Request-Line: ACK sip:[email protected]:5060;user=phone SIP/2.0
Message Header
Call-ID: 10
CSeq: 48 ACK
From: <sip:[email protected] :5060;user=phone>;tag=23E
Route: <sip:172.17.69.2:5060;lr>
To: <sip:[email protected];user=phone>;tag=1469
Via: SIP/2.0/UDP 192.100.100.2:5060;branch=z9hG4bK-5B7E-35
16- Src Addr: 172.17.69.2 (172.17.69.2), Dst Addr: 192.0.0.10 (192.0.0.10)
Request-Line: ACK sip:[email protected]:5060 SIP/2.0
Message Header
Call-ID: 10
CSeq: 48 ACK
From: <sip:[email protected];user=phone>;tag=23E
To: <sip:[email protected] :5060;user=phone>;tag=1469
Via: SIP/2.0/UDP 172.17.69.2:5060;branch=z9hG4bK6813179bf2588.1
Via: SIP/2.0/UDP 192.100.100.2:5060;branch=z9hG4bK-5B7E-35
Record-Route: <sip:172.17.69.2:5060;lr>
17- Src Addr: 192.100.100.2 (192.100.100.2), Dst Addr: 172.17.69.2 (172.17.69.2)
Request-Line: BYE sip:[email protected]:5060;user=phone SIP/2.0
18- Src Addr: 172.17.69.2 (172.17.69.2), Dst Addr: 192.0.0.10 (192.0.0.10)
Request-Line: BYE sip:[email protected]:5060 SIP/2.0
19- Src Addr: 192.0.0.10 (192.0.0.10), Dst Addr: 172.17.69.2 (172.17.69.2)
Status-Line: SIP/2.0 200 OK
20- Src Addr: 172.17.69.2 (172.17.69.2), Dst Addr: 192.100.100.2 (192.100.100.2)
Status-Line: SIP/2.0 200 OK
24
REFER SIP service transaction (1)
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Proxy/registrar server
@oneaccess-net.com
.69.2
172.17.68.0/22
.25
Sip phone
# 103
.69.1
REMOTE_SITE
.100
MAIN_SITE
.2
PABX BRI 5/0
IP network
.100
.10
192.0.0.0/24
BRI 5/0
BRI 5/1
PABX
192.100.100.0/24
#300
#0141877446
#301
25
REFER SIP service transaction (2)
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hostname MAIN_SITE
sip-gateway
reg-dns-add oneaccess-net.com
prox-dns-add oneaccess-net.com
gw-interface fastethernet 0/0 intrusive
bye-on-refer -> To support Refer SIP service
no shutdown
exit
Ip host oneaccess-net.com
dial-peer voice voip 0
sig-protocol sip
voip-coder-profile 0
no shutdown
exit
voice-routing
route 1
dial-peer pots-group 0 ua-sip
sip-authentication isdn_call1 300
prefix 300 length 3
prefix-type outgoing called last
exit
route 2
dial-peer pots-group 1 ua-sip
sip-authentication isdn_call2 O141877446
prefix 0141877446 length 10
prefix-type outgoing called last
exit
route 3
dial-peer voip 0
prefix . Timer -> this allows to route any other outgoing call to VoIP
prefix-type outgoing called last
exit
exit
26
REFER SIP service transaction (3)
Unattended Transfert global call flow
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
Phone # 301 calls Sip phone # 103 (who offhook). Then, SIP phone decides to transfert
(without consultation) the call to phone # 301 located at Remote_site
No.
1
2
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
Source
192.100.100.2
oneaccess-net.com
192.100.100.25
192.100.100.25
oneaccess-net.com
192.100.100.25
oneaccess-net.com
192.100.100.2
oneaccess-net.com
192.100.100.25
oneaccess-net.com
192.100.100.2
oneaccess-net.com
192.100.100.2
oneaccess-net.com
192.100.100.25
oneaccess-net.com
192.100.100.2
oneaccess-net.com
192.100.100.25
oneaccess-net.com
192.100.100.2
oneaccess-net.com
oneaccess-net.com
192.0.0.10
192.0.0.10
oneaccess-net.com
192.0.0.10
oneaccess-net.com
192.0.0.10
oneaccess-net.com
192.100.100.2
Destination
oneaccess-net.com
192.100.100.2
oneaccess-net.com
oneaccess-net.com
192.100.100.2
oneaccess-net.com
192.100.100.2
oneaccess-net.com
192.100.100.25
oneaccess-net.com
192.100.100.2
oneaccess-net.com
192.100.100.25
oneaccess-net.com
192.100.100.25
oneaccess-net.com
192.100.100.2
oneaccess-net.com
192.100.100.25
oneaccess-net.com
192.100.100.2
oneaccess-net.com
192.100.100.2
192.0.0.10
oneaccess-net.com
oneaccess-net.com
192.100.100.2
oneaccess-net.com
192.100.100.2
oneaccess-net.com
192.100.100.2
oneaccess-net.com
Info
Request: INVITE sip:[email protected]:5060;user=phone, with sdp
Status: 100 Trying
Status: 100 Trying
Status: 180 Ringing
Status: 180 Ringing
Status: 200 OK, with session description
Status: 200 OK, with session description
Request: ACK sip:[email protected]:5060;user=phone
Request: ACK sip:[email protected]
Request: REFER sip:[email protected]:5060;user=phone
Request: REFER sip:[email protected]:5060
Status: 202 ACCEPTED
Status: 202 ACCEPTED
Request: NOTIFY sip:[email protected]:5060;user=phone
Request: NOTIFY sip:[email protected], with Sipfrag(v=0...)
Status: 200 OK
Status: 200 OK
Request: BYE sip:[email protected]:5060;user=phone TO CLOSE THE FIRST CALL
Request: BYE sip:[email protected]
Status: 200 OK
Status: 200 OK
Request: INVITE sip:[email protected];user=phone, with sdp
Status: 100 Trying
Request: INVITE sip:[email protected]:5060, with sdp
Status: 100 trying
Status: 183 Proceeding
Status: 183 Proceeding
Status: 180 Ringing
Status: 180 Ringing
Status: 200 OK, with session description
Status: 200 OK, with session description
Request: ACK sip:[email protected]:5060;user=phone
34
oneaccess-net.com
192.0.0.10
Request: ACK sip:[email protected]:5060
27
REFER SIP service transaction (4)
Unattended Transfert, Refer/Notify method
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The next Messages are proxied by SIP proxy server (oneaccess-net.com,
IP address 172.17.69.2).
1- Message from 192.100.100.25 (SIP phone) to indicate a call transfert from
103 to 301(Unattended Transfert to phone 301).
REFER sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.17.69.2:5060;branch=z9hG4bKd9e6cb01abe12.3
Via: SIP/2.0/UDP 192.100.100.25;branch=z9hG4bKb15a9e22d64b4bc2
From: <sip:[email protected];user=phone>;tag=8059007111a3cdbb
To: <sip:[email protected]:5060;user=phone>;tag=3D0
Contact: <sip:[email protected]:5060;user=phone>
Supported: replaces
Refer-To: <sip:[email protected];user=phone> -> Form of Refer-To is like Contact header field
Referred-By: <sip:[email protected];user=phone> -> Form of Referred-By is like Contact
Call-ID: 12
header field
CSeq: 41316 REFER
User-Agent: Grandstream BT100 1.0.6.7
Max-Forwards: 69
Allow: INVITE,AC K,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Record-Route: <sip:172.17.69.2:5060;lr>
Content-Length: 0
28
REFER SIP service transaction (5)
Unattended Transfert, Refer/Notify method
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2- Message from 192.100.100.2 to acknowledge REFER request
SIP/2.0 202 ACCEPTED
Call-ID: 12
CSeq: 41316 REFER
From: <sip:[email protected];user=phone>;tag=8059007111a3cdbb
Record-Route: <sip: 172.17.69.2:5060;lr>
To: <sip:[email protected]:5060;user=phone>;tag=3D0
Via: SIP/2.0/UDP 172.17.69.2:5060;received=172.17.69.2;branch=z9hG4bKd9e6cb01abe12.3,SIP/2.0/UDP
192.100.100.25;branch=z9hG4bKb15a9e22d64b4bc2
Content-Length: 0
3- Message from 192.100.100.2 to inform « REFER » the sender of the status of
the reference (To, From, Call-ID)
NOTIFY sip:[email protected]:5060;user=phone
SIP/2.0
Call-ID: 12
Contact: <sip:[email protected]:5060;user=phone>
Content-Type: message/sipfrag -> this header field is mandatory (like that)
CSeq: 45 NOTIFY
Event: refer
-> Refer event package must be subscribed.
From: <sip:[email protected]:5060;user=phone>;tag=3D0
Route: <sip:172.17.69.2:5060;lr>
Subscription-State: terminated;reason=noresource -> This is the final Notify to the
To: <sip:[email protected];user=phone>;tag=8059007111a3cdbb
REFER Request
Via: SIP/2.0/UDP 192.100.100.2:5060;branch=z9hG4bK-7922
Content-Length: 150
SDP
29
Voice-routing: Backup scenarios (1)
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Proxy/registrar server
@oneaccess-net.com
3
.69.2
192.100.100.0/24
172.17.68.0/22
.69.1
REMOTE_SITE
.100
MAIN_SITE
.2
PABX BRI 5/0
BRI 5/1
IP network
2
1
.100
.10
192.0.0.0/24
BRI 5/2
BRI 5/0
BRI 5/2
PABX
#300
#0141877446
ISDN Network
#301
30
Voice-routing: Backup scenarios (2)
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
Based on the voice-routing configuration, this is
possible to re route any outgoing SIP call to an
alternate ISDN interface (ISDN call, not SIP)



This may be applied to backup the « uplink » which may
be a fast Ethernet, ATM, E1 interface.(1)
This may be applied to backup any failure in the IP
network, ATM pvc failure, routing…(2)
This may be applied to « backup » SIP server failure (3)
31
Voice-routing: Backup scenarios (3)
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CONFIGURATION :
voice-routing
route 1
dial-peer voip 2
prefix 014187300x
length 10
prefix-type outgoing called backup  if the primary route is not reachable
exit
Backup allows to jump to next entry
route 2
matching dialled prefix
dial-peer pots-group 0
prefix 0141873000
length 10
prefix-type outgoing called last
exit
.
.
route 4
dial-peer pots-group 2
prefix 0141873002
length 10
prefix-type outgoing called last
exit
32
Voice-routing: Backup scenarios (4)
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-Info vox voip controlplan 3 Incoming call on local port: 5/1, calling:
1001191101, called: 0141873002, call-id: 2.
-Error vox voip controlplan 1 Incoming call failure on local port: 5/1,
cause: (38)[Network out of order], call-id: 2.
-Info vox voip controlplan 3 Backup Process Dialing port: 5/1, number:
0141873002, call-id: 2
-Info vox voip controlplan 3 Outgoing call on local port: 5/2, calling:
1001191101,
called: 0141873002, call-id: 2.
-Info vox voip controlplan 3 Alert received, call-id: 2.
33
SIP Statistics
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
SIP Gateway Statistics
CLI# show voice sip-gateway
Sip-Gateway statistics :
Gateway state
Registration state
Registrar server
Bandwidth used/unused
Max Bandwidth exceeded
Registration errors
Registered endpoints
up
registered
192.168.1.100:5060
0 / 3072000 bps
0
0
2
Current call
0
Authentication Rejects
0
34
SIP Statistics
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
SIP Gateway Statistics
CLI# show voice sip-gateway endpoints
phone 1 :
State : registered
Number : 2408882000
SIP-id :
Registration timeout : 105s/120s
phone 2 :
State : registered
Number : 2408882001
SIP-id :
Registration timeout : 18s/120s
35
SIP Statistics
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
Events
CLI# event filter add vox voip all show
00:04:08.080 Info vox voip controlplan 3 FXS port 5/4 event:
off-hook, pots: 2.
00:04:12.580 Info vox voip controlplan 3 FXS port 5/4 event:
on-hook, pots: 2.
36
SIP Statistics
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
Debug
CLI#debug sip [registrar | signalling | all] - default: all
level [1..3] - default: 1 <cr>
CLI#debug sip registrar level 3
07:07:21.783 UAC [192.168.1.1:5060] -> UAS [192.168.1.100:5060]
REGISTER call-id:7057052324088820002C605143@oa-one100 (UDP)
REGISTER sip:192.168.1.100:5060;user=ip SIP/2.0
Call-ID: 7057052324088820002C605143@oa-one100
Contact: <sip:[email protected]:5060;user=phone>
CSeq: 124 REGISTER
Expires: 1800
From: <sip:[email protected]:5060;user=phone>;tag=401A
Max-Forwards: 70
Privacy: none
Supported: path
To: <sip:[email protected]:5060;user=phone>
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK-2D40-70
Content-Length: 0
CLI#debug sip registrar (about direction, method & call-id)
CLI#debug sip sig l 2 (about direction, method, call-id & sdp)
CLI#debug sip signalling level 3 (all details)
37
United Networking
Thank You for your
attention !
www.oneaccess-net.com
[email protected]
13 April, 2015
Copyright © OneAccess Networks – All rights reserved