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patton(cfg)#dns-lookup sip.voiceworks.pl
Name: sip.voiceworks.pl
Address: 192.168.1.2

debug gateway sip transport ;debug pełnych wiadomosci sip

patton(cfg)#08:58:43 SIP_TR> [GW] < Stack: from 192.168.0.4:40364
INVITE sip:111@192.168.0.68:5062;user=phone;transport=udp SIP/2.0
Record-Route: <sip:192.168.0.4:5080;lr>
Via: SIP/2.0/UDP 192.168.0.4:5080;branch=z9hG4bK-df0e05c7a4068097f35fcdb274f52062
Via: SIP/2.0/UDP 192.168.0.4;branch=z9hG4bK-6d73755911e4379b62331d1fbe6fba4d
Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK55d02100CF305BD5
From: "Lukasz Kutkowski" <sip:305@sip.voiceworks.pl>;tag=3BF55A5A-6E4CDC71
To: <sip:550@sip.voiceworks.pl;user=phone>
Cseq: 1 INVITE
Call-Id: dd18ee8e-98d1532c-4981f7fb@192.168.0.253
Contact: <sip:305@192.168.0.253>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_300-UA/2.1.0.2708
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Max-Forwards: 18
Content-Type: application/sdp
Content-Length: 251
Date: Mon, 29 Oct 2007 08:58:44 GMT

v=0
o=- 1193648255 1193648255 IN IP4 192.168.0.253
s=Polycom IP Phone
c=IN IP4 192.168.0.253
t=0 0
m=audio 2234 RTP/AVP 18 8 0 101
a=sendrecv
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000

debug gateway sip transport detail 1 ;pokazuje tylko rodzaj i kody wiadomości SIP

patton(cfg)#08:56:34 SIP_TR> [GW] > Stack: SIP/2.0 503 Service Unavailable
08:56:34 SIP_TR> [GW] < Stack: INVITE sip:111@10.2.250.68:5062;user=phone;transport=udp SIP/2.0
08:56:34 SIP_TR> [GW] > Stack: SIP/2.0 100 Trying
08:56:34 SIP_TR> [GW] > Stack: SIP/2.0 503 Service Unavailable

debug media-gateway rtp ;pozwala na śledzenie połączenia na poziomie rtp

17:42:19 RTP > [02000021] (BCD) Event=rx-rtp | New State=established
17:42:20 RTP > [0200001f] RTCP TX
17:42:20 RTP > [0200001f] RTCP-TX: TX-Info: packets=86, octets=1720
17:42:20 RTP > [0200001f] RTCP TX: RX-Info: packets=78, octets=1560, lost=0, jitter=14
17:42:20 RTP > [02000022] RTCP TX
17:42:20 RTP > [02000022] RTCP-TX: TX-Info: packets=76, octets=1520
17:42:20 RTP > [02000022] RTCP TX: RX-Info: packets=96, octets=1920, lost=0, jitter=0
17:42:20 RTP > [02000021] RTCP TX
17:42:20 RTP > [02000021] RTCP-TX: TX-Info: packets=138, octets=22080
17:42:20 RTP > [02000021] RTCP TX: RX-Info: packets=76, octets=12160, lost=0, jitter=0
17:42:22 RTP > [0200001f] Set mode: TX/RX -> INACTIVE
17:42:22 RTP > [0200001f] (BCD) Event=disable | New State=disabled

debug isdn event <slot> <port> {all | layer2 | layer3} ;pozwala na debug warstwy 2 i 3 dla portu BRI

17:44:59 ISDN > # 148 p: 0 R: sapi: 0 cr=0 ea=0 tei: 66 ea=1 RR Nr( 1) pf=0
17:44:59 ISDN > # 149 p: 0 R: sapi: 0 cr=0 ea=0 tei: 67 ea=1 RR Nr( 1) pf=0
17:44:59 ISDN > # 150 p: 0 from CC key: 259 L3ProceedingReq
17:44:59 ISDN > # 151 cause : Transparent
17:44:59 ISDN > # 152 hex:1E 02 85 82
17:44:59 ISDN > # 153 IE: ProgressInd
17:44:59 ISDN > # 154 ---> Layer2 tei:127 sapi: 0 CallProc
17:44:59 ISDN > # 155 hex:08 01 FC 02 18 01 89 1E 02 85 82
17:44:59 ISDN > # 156 IE: ChannelId BCh0 excl otherIf
17:44:59 ISDN > # 157 IE: ProgressInd
17:44:59 ISDN > # 158 new call state : L3PcSt09U
17:44:59 ISDN > # 159 p: 0 S: sapi: 0 cr=0 ea=0 tei: 76 ea=1 SABME p =1
17:44:59 ISDN > # 160 p: 0 R: sapi: 0 cr=0 ea=0 tei: 76 ea=1 UA f =1
17:44:59 ISDN > # 161 p: 0 S: sapi: 0 cr=0 ea=0 tei: 76 ea=1 INFO Nr( 0) Ns( 0) pf=0


show call-control call ;pokazuje aktualny stan połączeń

Active Calls
============
Call Call-Leg State Address Display Charge
-----------------------------------------------------------------------------------------------

00919e60
IF-ISDN-TPSA-00e5484 CONNECTED 90228753293 n/a n/a
VIA.PINGTEL-4726003e CONNECTED 100 n/a n/a
010cef70
VIA.PINGTEL-4726003e CONNECTED 90228753293 n/a n/a
IF-SIP-PINGTEL-010ce CONNECTED 100 n/a n/a

show call-control call continuously ;pokazuje aktualny stan połączeń odświeżając co kilka sekund

debug call-router ;pokazuje jak została kierowana rozmowa w CS podczas zestawiania połączenia

patton#15:47:30 CR > [switch] Routing-Lookup:
15:47:30 CR > Execute all entries in table interfejs_sip-precall-service
15:47:30 CR > Execute all entries in table interfejs_PRI_wejscie-dest
15:47:30 CR > Execute all entries in table route-found-place-call
15:47:30 CR > Lookup result: Route found; place call (timeout=0)

debug call-control ;pokazuje jakie parametry ma rozmowa (skąd przyszła, dokąd idzie, kody rozł)

patton#15:48:24 CC > [EP interfejs_sip-013dd410/active] Set call-leg property: Network -> SIP-GW
15:48:24 CC > [EP interfejs_sip-013dd410/active] Set call-leg property: E164-Number -> 600
15:48:24 CC > [EP interfejs_sip-013dd410/active] Set call-leg property: URI -> sip:600@sip.voiceworks.pl
15:48:24 CC > [EP interfejs_sip-013dd410/active] Set call-leg property: Type-Of-Number -> Unknown
15:48:24 CC > [EP interfejs_sip-013dd410/active] Set call-leg property: Numbering-Plan -> Unknown
15:48:24 CC > [EP interfejs_sip-013dd410/active] Set call-leg property: Presentation-Indicator -> Presentation allowed
15:48:24 CC > [EP interfejs_sip-013dd410/active] Set call-leg property: Name -> Lukasz
15:48:24 CC > [EP interfejs_sip-013dd410/active] Set call-leg property: Screening-Indicator -> User provided, not screened
15:48:24 CC > [EP interfejs_sip-013dd410/active] Set call-leg property: Supports Overlap-Sending -> true
15:48:24 CC > [EP interfejs_sip-013dd410/active] Set call-leg property: Supported Codecs -> Voice: G.711 A-law[20/20], G.711 u-law[20/20]
15:48:24 CC > [EP interfejs_sip-013dd410/active] Set call-leg property: Unique Identifier -> a64279de-2a172134-79d06ce1@192.168.0.253
15:48:24 CC > [EP interfejs_sip-013dd410/active] Set call-leg property: IP-Address -> 192.168.0.253
15:48:24 CC > [EP interfejs_sip-013dd410/active] Set call-leg property: Call-Leg-ID -> 0x013dfd58
15:48:24 CC > [EP interfejs_sip-013dd410/active] Set call-leg property: State -> CONNECTED
15:48:24 CC > [Call 013e0400] Set call property: Context -> 0x00000000
15:48:24 CC > [Call 013e0400] Set call property: Information-Transfer-Capability -> 3.1kHz Audio

show port <ethernet | bri | isdn > <port number> ;pokazuje status portu podanego jako parametr

patton#show port e1t1 0 0
Port: e1t1 0 0 0
================

Clock: Auto
Higher layer clock preference: Slave
Line Code: HDB3
Framing: Multiframe with CRC4

IDT 82P228x
===========

Type: E1
Port State: Enabled
Link State: Wait-Signal
Clock: Slave
Line Code: HDB3
Framing: CRC4
Loopback: Disabled
Application: Short-Haul
CAS channels: 0
Red alarm: On
AIS alarm: Off
Basic Framing: no
Multi Framing: no
Crc4 to Non-Crc4 Interworking: no
Bit Slips (local clock too fast): 0
Bit Slips (local clock too slow): 1
Rx Frames: 362
Rx Bytes: 333723
Tx Frames: 362
Tx Bytes: 1560
Impedance: 120 Ohm

no debug all ;wyłącz logi

show log ;podstawowy log pokazujący np. stany łączy

2007-10-29T08:45:51 : LOGINFO : Link down on interface e1t1 0 0 1
2007-10-29T08:45:51 : LOGINFO : Link down on interface ISDN 0 1
2007-10-29T08:46:55 : LOGINFO : Link up on interface e1t1 0 0 1
2007-10-29T08:46:55 : LOGINFO : Link up on interface ISDN 0 1
2007-10-29T14:32:03 : LOGINFO : Link down on interface ethernet 0 0 1
2007-10-29T14:32:16 : LOGINFO : Link up on interface ethernet 0 0 1

show license ;pokazuje zainstalowane licencje

patton#show licenses

IP Routing [iprouter]
Status: Active

VPN [vpn]
Status: Active

Sponsored by: 

Poniżej przedstawiona w sposób graficzny dozwolone konfiguracje SIP w bramkach SmartNode z firmwarem 3.X oraz 4.X. Numerki na rysunkach sugerują kolejność w jakiej powinny być konfigurowane poszczególne elementy niezbędne dla działania interfejsu SIP.

W celu sprawdzenia wersji firmwaru naleźy w wierszu poleceń wywołać koemndę: show version 

Sponsored by:

Konfiguracja routingu rozmów pomiędzy różnymi technologiami

1. Interface IP (IF IP) – podstawowym parametrem jest adres IP i maska podsieci które określają sieć/podsieć w jakiej będzie widoczny ten interfejs.

Wejście do trybu konfiguracji przykładowego interfejsu IP:

telnet patton

podać login i hasło

patton>enable

patton#configure

patton(cfg)#context ip router

patton(ctx-ip)[router]#interface LAN

patton(if-ip)[IF-IP-L~]#ipaddress 192.168.1.1 255.255.255.0

patton(if-ip)[IF-IP-L~]#

Naciskając TAB pojawią się komendy konfiguracyjne dla tego interfejsu.

Konfiguracja:

contex ip router

interface WAN

ipaddress dhcp

use profile napt NAPT_WAN

tcp adjust-mss rx mtu

tcp adjust-mss tx mtu

interface LAN

ipaddress 192.168.1.1 255.255.255.0

tcp adjust-mss rx mtu

tcp adjust-mss tx mtu

2. Port Ethernet (port eth) – jest to fizyczna konfiguracja portu np. 10Mb/s czy 100Mb/s itp.

patton(if-ip)[IF-IP-L~]#end

patton#

patton#configure

patton(cfg)#port ethernet 0 1

patton(prt-eth)[0/1]# bind interface LAN router

patton(prt-eth)[0/1]#

Naciskając TAB pojawią się komendy konfiguracyjne dla tego interfejsu.

Konfiguracja:

port ethernet 0 1

medium auto

encapsulation ip

bind interface LAN router

no shutdown

 

3. Gateway SIP (GW SIP) – konfiguracja podstawowych parametrów SIP jak port (jeśli ma być inny niż domyślny 5060), domain, realm itp.

patton(prt-eth)[0/1]#end

patton#

patton#configure

patton(cfg)#gateway sip SIPGW-VOIP

patton(gw-sip)[SIPGW-V~]#bind interface LAN router

patton(gw-sip)[SIPGW-V~]#service default

patton(svc-sip)[default]#domain voiceworks.pl

patton(svc-sip)[default]#

Konfiguracja:

gateway sip SIPGW-VOIP

bind interface LAN router

service default

domain voiceworks.pl

gateway sip SIPGW-VOIP

no shutdown

 

4. Interface SIP (IF SIP) – interfejs ten daje dostęp do contextu cs który pozwala na kierowanie rozmowy oraz modyfikacje jej parametrów. W interfejsie są konfigurowane parametry specyficzne dla protokołu SIP.

patton(svc-sip)[default]#end

patton#

patton#configure

patton(cfg)#context cs switch

patton(ctx-cs)[switch]#interface sip IF-SIP-VOIP

patton(if-sip)[IF-SIP-~]#bind gateway SIPGW-VOIP

patton(if-sip)[IF-SIP-~]#remote sip.voiceworks.pl

patton(if-sip)[IF-SIP-~]#service default

patton(if-sip)[IF-SIP-~]#route call dest-table TAB-FROM-VOIP

route call - dotyczy połączeń które przychodzą na interfejs i są przekazywane do contextu CS.

Konfiguracja:

context cs switch

interface sip IF-SIP-VOIP

service default

route call dest-table TAB-FROM-VOIP

remote sip.voiceworks.pl

context cs switch

no shutdown

 

Konfiguracja interfejsu BRI

5. Interface ISDN (IF BRI) – interfejs ten daje dostęp do contextu cs który pozwala na kierowanie rozmowy oraz modyfikacje jej parametrów. W interfejsie są konfigurowane parametry specyficzne dla sieci ISDN (protokołu DSS1).

patton(if-sip)[IF-SIP-~]#end

patton#

patton#configure

patton(cfg)#context cs switch

patton(ctx-cs)[switch]#interface isdn IF-ISDN-TELCO

patton(if-isdn)[IF-ISDN-~]#route call dest-table TAB-FROM-ISDN

route call - dotyczy połączeń które przychodzą na interfejs i są przekazywane do contextu CS.

Konfiguracja:

context cs switch

interface isdn IF-ISDN-TELCO

route call dest-table TAB-FROM-ISDN

interface sip IF-SIP-VOIP

service default

route call dest-table TAB-FROM-VOIP

remote sip.voiceworks.pl

context cs switch

no shutdown

 

6. Port BRI (port bri) – jest to fizyczna konfiguracja portu BRI.

patton(if-isdn)[IF-ISDN~]#end

patton#configure

patton(cfg)#port bri 0 0

patton(prt-bri)[0/0]#

patton(prt-bri)[0/0]#q921

patton(q921)[0/0]#encapsulation q931

patton(q921)[0/0]#uni-side auto

patton(q921)[0/0]#q931

patton(q931)[0/0]#encapsulation cc-isdn

patton(q931)[0/0]#protocol dss1

patton(q931)[0/0]#uni-side user

patton(q931)[0/0]#bind interface IF-ISDN-TELCO switch

patton(q931)[0/0]#exit

patton(q921)[0/0]#exit

patton(prt-bri)[0/0]#no shutdown

Konfiguracja:

port bri 0 0

clock auto

encapsulation q921

q921

protocol pmp

uni-side auto

encapsulation q931

q931

protocol dss1

uni-side user

encapsulation cc-isdn

bind interface IF-ISDN-TELCO switch

port bri 0 0

no shutdown

 

Context CS

1.Context cs służy do:

• kierowania rozmowy,
• modyfikacja parametrów rozmowy np. zmiana numeru prezentowanego itp.

2.Dostęp do contextu cs jest tylko i wyłącznie poprzez skonfigurowany interfejs (np. IF SIP,IF BRI),

3.Elementem docelowym każdej rozmowy w context cs jest interfejs,

4.Każdy interfejs musi mieć określone gdzie dalej ma zostać kierowana rozmowa wchodząca do contextu.

patton(prt-bri)[0/0]#end

patton#

patton#configure

patton(cfg)#context cs switch

patton(ctx-cs)[switch]#routing-table called-e164 TAB-FROM-VOIP

patton(rt-tab)[TAB-FRO~]#route default dest-interface IF-ISDN-TELCO

patton(rt-tab)[TAB-FRO~]#exit

patton(ctx-cs)[switch]#routing-table called-e164 TAB-FROM-ISDN

patton(rt-tab)[TAB-FRO~]#route default dest-interface IF-SIP-VOIP

Konfiguracja:

context cs switch

routing-table called-e164 TAB-FROM-VOIP

route default dest-interface IF-ISDN-TELCO

routing-table called-e164 TAB-FROM-ISDN

route default dest-interface IF-SIP-VOIP

interface isdn IF-ISDN-TELCO

route call dest-table TAB-FROM-ISDN

interface sip IF-SIP-VOIP

service default

route call dest-table TAB-FROM-VOIP

remote sip.voiceworks.pl

 

context cs switch

no shutdown

Performance tests of 16E1s SS7 Sangoma Media Gateway

Sponsored by:

Abstract:

In this article we present performance data on 16E1 SS7 setup based on Sangoma's SMG & Asterisk as a voice application server.
CPU usage and Unix system load are analysed as a function of number of calls in the answered state both for centralized & distributed setup. Load analysis shows that in a distributed setup a Dual Core 3GHz AMD Opteron server has enough resources to handle full 16E1 load with room for additional applications like SIP termination.

Performance of 8E1s Sangoma Netborder Express Gateway

Sponsored by:

Required Level: novice, accomplished, adept, master

Abstract:

Sangoma's NetBorder Express is a complete SIP-compliant VoIP to TDM Media Gateway solution that is user friendly and works under Windows operating system . Product consist of Sangoma A10(1/2/4/8) cards (max  32xE1) and dedicated software+drivers.  In the article we present results of performance test of this product (8xE1 setup) with FreeSWITCH as a SIP terminating PBX. Tests took place in VoiceWorks laboratory. 

Goals:

Test CPU and RAM memory usage in two scenarios :

• 1 machine - FreeSWITCH and Netborder Express Gateway on the same server,
• 2 machines - Netborder on Windows machine and FreeSWITCH on Centos Linux

Purpose of this test is to measure performance of the latest Sangoma's product under Windows platform.

Lab Setup :

• Hardware
  
  
  • HP DX2400,  Intel Core 2 Quad CPU Q9300 (4x2,5 GHz), 2 GB RAM, 250 GB SATA HDD (Windows)
  • noname Intel Xeon 2x2 GHz, 2 GB RAM, 2x110 HDD (Centos Linux)
  • Sangoma A108D PCI-e cards, 8 E1 port PSTN cards with HW Echo Cancellation
  • 2 x Patton 4960 Gateways (4xE1 each)
  • Netborder Express Gateway Card

• Sofware
  
  
  • Windows XP Professional Service Pack 3
  • Netborder Express Gateway v 1.6.1
  • Sangoma Windows Driver 6.0.5.8
  • CentOS 5.2
  • FreeSWITCH version 1.0.trunk

Methodology:

Call scenarios for both setups are the same. We start with empty system and add a new call every 3 seconds untill all chanells are occupied. Then during the period of 5 minutes we drop a random call (every 375 ms) and than replace it with the a new one. Below there is the description of a single call flow:

1. G.711 SIP call is originated from FreeSWITCH to Sangoma's Netborder Express,
2. Netbord forwards the call to Patton,
3. Patton loops the call back to Netboard,
4. Netbord forwards the call back to FreeSWITCH,
5. FreeSWITCH answers the call and exectues "Echo" application.

Basically every call occupies 2 time slots.

During the tests we monitor CPU utilization (Total and for every Core), memory usage and HD activity. All the parameters on Windows machine are monitored by build in Performance Monitor. To achieve maximum performance  we are disabling all debugs on the Netboard machine.

Patton boxes are also used to check if Sangoma's new products is fully compliant with Euro-ISDN standards.  

Configuration:

 FreeSWITCH config files:

• default.xml (dialplan)
• sofia.conf.xml
• switch.conf.xml

 Netboard Express Gateway config files:

• routing-rules.xml

 

Results:

Conclusions:

Tests showed that there is a lot of spare resources for increasing capacity of the system or building complex logic when FreeSWITCH is running under Windows even on everyday desktop computer. Both setups are able to easily handle 8E1s and uses less than 1/4 of the CPU. Simple extrapolation confirms that distributed setup could handle at least 32xE1 on a single machine. CPU load is evenly distributed between cores which guarantees optimal resources usage. 

This lab setup also hits at solution for all Open-Source integrators that are faced with obstacles like Windows environment or strict ISDN compliance requirements for their installs.

Sangoma's Netborder Express comes with certification from huge number of vendors opening doors to every possible tender. When coupled with FreeSwitch, the ultimate Open-Source telephony integration tool, running along on the same Windows machine one gets an unbeatable combo in the market segment previously reserved only for selected players.