Title: Voice
1(No Transcript)
2Voice
3Overview
- Introduction
- SIP Basic Configuration
- SIP Advanced Configuration
- SIP Services
- Codec Use
- Country Specific Settings
- Advanced Voice Routing
- Debug VoIP
4Introduction
5SIP Network Elements
Proxy
SIP phone
IP Network
Registrar
IP / PSTN Gateway
PSTN phone
SIP phone
SIP phone
6SIP Session Setup Example
SIP User Agent Client
SIP Proxy
INVITE sippicard_at_uunet.com
200 OK
ACK
Media Stream
BYE
200 OK
host.wcom.com
sip.uunet.com
7SIP Protocol Main Actors
- Client board, SoftPhone, SIP phone
- Board / Voice ports
- FXS1 / FXS2
- DECT
- SIP phone ST2020, ST2030, ST288
- Registrar
- Proxy
- Gateways
8VoIP on Board
- Inbound SIP client
- Voice Codecs
- Voice Services
- Others
9SIP ALG
- Handles connections
- Does NAT mapping
- Replaces parts of SIP message containing address
/ port of Local Network to corresponding ones of
NAPT mapping
10Inbound SIP Client Illustrated
11SIP Basic Configuration
12Basic Configuration GUI Access
- Factory defaults on LAN
- DHCP-Server enabled on PC
- Thomson Gateway IP-address 192.168.1.254
- URL http//192.168.1.254
- Telnet/GUI Username Administrator
- No password
- Thomson Gateway Set-Up Wizard
- This launches pop-up window for easy configuration
13Toolbox Telephony
14Toolbox Telephony Expert Configure
- Registrar used as domain-name
- FQDN or IP
- Proxy SIP destination Server IP-address
- FQDN or IP
15Toolbox Telephony Configure
- SIP URI telephone number
- Username and Password SIP authentication
- Display name CLIP
- Abbr. Number internal number between local
phones - Port assign number to
16SIP Registration
- VoIP activation
- Enable Telephony
- Register successful
- Register not successful
17SIP Advanced Configuration
18service System
- Service manager on Thomson Gateway
- Enable / disable service
- Change local port (source port)
- Access list on Interface/IP-address
- Labels for routing and QoS
- Service manager automatically generates
- NAT entries
- Firewall rules
19service System Continued
- Manually enable VoIP serviceservice system
modify nameVOIP_SIP stateenableservice system
list expandenablede.g. Change voice source
port, default5060service system modify
nameVOIP_SIP port5091
206.2 SIP-ALG
- Since 6.2, SIP-ALG has to be used for local voice
application - gtconnection bindlist
- Application Proto Portrange Flags
- SIP udp 5060 SIP_ALGE
RTP_predict_for_term_SIP_ALGE - if not in list
- connection bind applicationSIP port5060
- if FLAG not set
- automatically learning RTP flow from SIP SDP
negotiation - connection appconfig applicationSIP
RTP_predict_for_term_SIP_ALGenabled
21Voice Ports
- FXS Analogue phone
- FXO PSTN-line
- DECT DECT handsets associated to ST7x7
- Common ports are groups of voice ports
- All / COMMON all available ports
- All DECT
22Common Number
- Multiple common numbers supported
- Outgoing call from local voice port uses COMMON
number as CID (Caller ID) - When no local number specifically configured for
that port - Automatically activated when COMMON number used
- Cannot be withdrawn
- Incoming call on COMMON number rings all
available ports
23Voice HW Regulatory and FXO
- Analogue Outgoing Telephone LinePOTS Back-Up
- Level 0 No FXO nor PSTN Backup
- Level 1 ONLY support for power failure to dial
out - Level 2 Power Failed VoIP Service Failed
Prefix dialing first then PSTN Incoming FXO
Calls - Reduced FXO
- Level 3 Power Failed VoIP Service Failed
Prefix dialing first then PSTN Incoming FXO
Calls Emergence call without prefix dialing
needs (ex 110, 911... Specific No. can be
configurable by customers...etc.) - Full FXO
706 / 716
780 / 785
24Voice FXO
- gtvoice fxoport config
- Incoming fxo enabled
- FXO disconnect timer 1000
- incfxo ltenableddisabledgt
- Enable or disable incoming FXO calls ?
disable FXO relay - fxodisconnect ltnumber500-5000gt
- The FXO disconnect timer (in ms)
25voice Configuration
- CLI gtvoice help config
- autofxo ltdisabledenabledgt automatically
make FXO calls when not registered - digitrelay ltautoinbandrfc2833signallinggt
set digit relay mode - click2dial_ports ltFXS1FXS2allgt set click
to dial port - rtp_portrange ltport-rangegt RTP port range
- sign_internal ltexternalinternalgt
signalling for local calls kept local or external - static_intf ltdisabledenabledgt use static
(configured) interface to look for source IP
address or not - intf ltloopInternetLocalNetworkgt name of
IP interface used for VOIP traffic
26voice Configuration Continued
- CLI gtvoice help config
- secondintf ltloopInternetLocalNetworkgt
name of backup IP interface used for VOIP traffic - endofnumber ltgt end of number character
for dialled number starting with cipher - countrycode ltnumber0-999gt local country
code - delayeddisconnect ltenableddisabledgt
enable or disable delayed disconnect feature - delayeddisconnecttimer ltnumber1-600gt
delayed disconnect timer (in seconds) - ringmuteduration ltnumber0-60000gt early
media mute duration (in minutes)
27SIP Configuration Overview
- voice sip config
- UserAgent domain
thomsongateway.sip - Primary proxy address
voip.thomson.be5060 - Secondary proxy address 0.0.0.05060 ?
Not supported - Primary registrar address voip.Thomson
Gateway.be5060 - Secondary registrar address 0.0.0.05060 ?
Not supported - Listening port 5060
- Expire time 3600
- Expire time delta 1
- Notifier address 0.0.0.05060
- Subscribe expire time 3600
28SIP Configuration Overview Continued
- voice sip config
- Call Waiting reply 182
- Transport UDP
- rtpmapstaticPT Disabled
- reinvite_stop_audio Disabled
- PRACK Disabled
- Clir format standard
- DTMF / in INFO method 1011
- Clip consider displayname yes
- SDP packet time 20
- Replace Enabled
- Symmetric codec Enabled
- Reinvite at calling fax detect Disabled
- SIPURI port Enabled
- rport Disabled
- SDP username 780
- ringtoneat183 Disabled
- T38 Port increment 0
29voice Profile Create VoIP User
- Voice profile add
- SIP_URI ltstringgt ? telephone number
- SIP URI related to voice port username
ltstringgt - Authentication username related to voice
portpassword ltpasswordgt - Authentication password related to voice
portdisplayname ltstringgt ? CLIP info - Alias name for SIP_URIvoiceport
ltFXS1FXS2COMMONgt ? available ports on TG - Analogue line number abbr ltstringgt
- Abbreviated number mapped to SIP_URI
- abbr. number only supported when URI has NO
LETTERS, only numbers - gtvoice profile list SIP_URI all
- Port Uri DisplayName Username
Abbr Nbr RegStatus Msg Waiting - --------------------------------------------------
------------------------------------------- - COMMON 2585 2585 2585
85 Registered No
30voice Country
- gtvoice country config countrybelgium
- Pre-loaded country settings
- Country australia belgium denmark etsi
france1 france2 france3 germany italy
netherlands northamerica norway spain
sweden uk - Country specific settings
- DTMF tones / dial tones / Hook flash timer /
polarity / etc. - Can be changed to specific needs
- Special file in dl-directory vincfg.bin
31voice Cac
- gtvoice cac help config
- maxportsperprofile ltoneallgt
- Maximum number of ports that can be used with
common profile - One only 1 call with common number possible at
same time - All multiple calls with common number possible
at same time
32SIP Services
33Local Services 3 Port Call
- What is needed for 3-way conference call using
VoIP?
- Answer or make call
- Put 1st call on hold
- Make 2nd call
- Switch between calls or put in 3-way conference
call
Services available?
Voice Network
hold R2
3way R3
switch R2
34Supplementary Services SpeedTouch 6.1
- Transfer Call Transfer between local ports
- Hold put active Call on Hold
- Waiting incoming call while active call
indication - Mwi Message Waiting Indication
- Clip Calling Line Identification Presentation
- Clir Calling Line Identification Restriction
- 3pty Three Party Call
35Supplementary Services SpeedTouch 6.1 Continued
- forcedFXO switch to FXO (PSTN)
- Cfu Call Forwarding Unconditional
- Cfnr Call Forwarding on No Reply
- Cfbs Call Forwarding on Busy
- Ccbs Call Completion on Busy Subscriber
- Clironcall CLIR for only one call
- Waitingoncall Call Waiting active for only one
call
36Codec Use
37Codec Support
Codec
G.711 audio at 64 Kbit/s, A-law
G.711 audio at 64 Kbit/s, µ-law
G.723.I at either 5.3 or 6.3 Kbit/s
G.723.I at either 5.3 or 6.3 Kbit/s with silence suppression as in AnnexA
G.726 ADPCM at 16 Kbit/s
G.726 ADPCM at 24 Kbit/s
G.726 ADPCM at 32 Kbit/s
G.726 ADPCM at 40 Kbit/s
G.729AnnexA audio at 8 Kbit/s
G.729AnnexA audio at 8 Kbit/s with silence suppression as in AnnexB
38Advanced Voice Routing
39Advanced Routed Scenario
- Multiple routed interfaces
- How VoIP interface binding?
- VoIP routing-based
- Best route used
- SIP SoftSwitch is known IP-address
- UA and RTP?
- No IP-address known
- All VoIP GW configured in routing ? complex
40Internal Overview
VoIP - SIP application
Routing
dhcp voice_ip_intf
PPP
ip voice_ip_intf
PPP relay
PPP - default MAC
MER - USB MAC
Eth-mer-if
Eth bridge
default bridge
Eth-mer-if is connected to bridge group video
ETH ports
video group id2
bridge pvc 8/35
Flexiport move Eth-port to video group
41VoIP Interface Specific
- Source IP address selection (on interface)
- voip config static_intfenabled
intfvoip_ip_intf - VoIP label for RTP routing
- RTP via SIP-ALG and label inheritance
- (default 7.4)
- gtconnection bindlist
- Application Proto Portrange Flags
- SIP udp 5060 SIP_ALGE
RTP_predict_for_term_SIP_ALGE
Administratorlabelgtlist Name Class
Def Ack Bidirect Inherit
Tosmark Type Value Use Trace
-------------------------------------------------
--------------------------------------------------
-------- DSCP overwrite dscp
prioritize disabled disabled disabled tos
0 1 disabled Interactive increase 8
6 disabled disabled disabled
tos 0 14 disabled Management
increase 12 12 disabled
disabled disabled tos 0 4
disabled Video increase 10 10
disabled disabled disabled tos 0
2 disabled VoIP-RTP overwrite 14
14 enabled disabled disabled tos
0 1 disabled VoIP-Signal overwrite
12 12 enabled disabled disabled
tos 0 2 disabled default
increase default prioritize disabled
disabled disabled tos 0 1
disabled
42Add Label Routing
- default QoS rule
- label rule add chainqos_default_labels index3
servsip logdisabled stateenabled labelVoIP - to be added for voice routing
- label add Voip_only inheritanceenabled
- label rule add chainrt_user_labels index1
srcintflocal servsip logdisabled stateenabled
labelVoip_only - Label rule instructs that all CPE SIP traffic has
to use this label
43Label Routing / Forwarding
- Static IP-address ip rtadd dst 0.0.0.0/0
labelVoip_only gateway123.123.123.123 - PPP VoIP interfaceppp rtadd intfInternet
dst0.0.0.0/0 labelVoip_only - DHCP-Client VoIP interfacedhcp client ifconfig
intfvoip_ip_intf labelVoip_only gatewayenabled
44Debug VoIP
45Tips and Tricks
- Services not working
- No hookflash detection etc.?
- Change country to etsi or northamerica
- Different analogue phone settings / timers
- One way voice
- Codec priority changes
- Enable / disable codecs
- Ptime changes
- Ptime acceptable for Gateway?
- Connection bindlist
- SIP-ALG bounded?
- Symmetrical codec
- Enabled for 7G?
46Debug VoIP Standard Traces
- Ctrl-q ? start debug
- Ctrl-s ? stop debug
- Ctrl-t ? clear buffer
- Ethereal trace has VoIP flow
- Statistics VoIP Call
- Additional voice traces
- Enable ? voice debug exec cmdtrace 1
- Disable ? voice debug exec cmdtrace 0
47Hands on - debugging
- Install and configure x-lite (SoftPhone)
48Debugging
49Debugging
50Debugging
- 200 OK is received by the CPE but not forwarded
to the Computer running x-lite
51Debugging
- INLocalNet-gt 192.168.1.64
192.168.0.101 0583 UDP 62748-gt5080 - UTLocalNet-gtip_voice 192.168.0.104
192.168.0.101 0583 UDP 49252-gt5080 - IN ip_voice-gt 192.168.0.101
192.168.0.104 0443 UDP 5080-gt62748 - DR ip_voice-gtip_voice 192.168.0.101
192.168.0.104 0443 UDP 5080-gt62748 - error caused by NAT-INPUT
- The 200 OK is dropped because the packet is
received on the wrong port -
- Why does the SIPServer reply on 62748?
52Solving
- The data sent to the SIPserver is wrong obviously
because of NAT. - The ALG should modify the content and generate
the appropriated NAT entry. - The SIP ALG is bound to the wrong Port
Administratorgtconnection bindlist
Application Proto Portrange Flags SIP
udp 5060 SIP_ALGE
RTP_predict_for_term_SIP_ALGE IKE udp
500
Administratorgtconnection bind applicationSIP
port5080