Asterisk SIP Settings

Asterisk SIP Settings
Asterisk SIP Settings
User Guide
Schmooze Com Inc.
Asterisk SIP Settings
User Guide
Chapters
Overview
Logging In
NAT Settings
Audio Codecs
Video Codecs
Media & RTP Settings
Notification & MWI
Registration Settings
Jitter Buffer Settings
Advanced General Settings
Recap
© Copyright Schmooze Com, 2013
Asterisk SIP Settings
User Guide
Overview
It was necessary in the past to hand edit files like “/etc/asterisk/sip_nat.conf” as part of the initial
installation of any Asterisk based deployment. With the introduction of the Asterisk SIP Settings
module, most SIP settings are made available in the GUI. This includes the all important NAT,
External IP, Local Network, Enabled Codecs and Codec order.
Logging In
• Log into the Asterisk SIP Settings module and you should see a screen like this.
Edit Settings
NAT Settings
NAT
yes
IP Configuration
Public IP
no
External IP
12.144.14.19
Local Networks
192.168.1.0
never
route
Static IP
Auto Configure
Dynamic IP
/ 255.255.255.0
Add Local Network Field
NAT Settings
• NAT settings are crucial in avoiding issues with one-way audio. You may need to adjust these
settings to match the network environment in which you place your system.
◦ NAT- (Network Address Translation) The method used to pass network traffic between the
public and private IP networks. We recommend setting this to “Yes” if you are going to have
SIP trunks or remote phones on a different network then the PBX.
▪
▪
▪
▪
Yes Mode- This will always ignore info and assume NAT.
No Mode- This will use NAT only according to RFC3581.
Never Mode- This will never attempt to use NAT mode or RFC3581.
Route Mode- This will assume NAT and not send rport.
◦ IP Configuration- Indicate whether the system has a public IP or requires NAT setting.
Automatic configuration of what is often set as “sip_nat.conf.”
◦ External IP
▪ (Set as static IP - asterisk:externip)- External static IP or FQDN as seen on the WAN
side of the router.
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Asterisk SIP Settings
User Guide
▪ (Set as dynamic IP - asterisk:externhost)- External FQDN as seen on the WAN side of
the router and updated dynamically, i.e. mydomain.dyndns.com.
◦ Local Networks
▪ (asterisk:localnet)- Local network settings in the form of “ip/mask” such as,
“192.168.1.0/255.255.255.0.” For networks with more than one LAN subnet, such as
VPN network, us the “Add Local Network” button to add more fields. Blank fields will be
removed upon submitting.
Audio Codecs
• Check the desired codecs and drag to reorder. All others will be disabled unless explicitly
enabled in a device or trunk configuration. Note that some codecs, such as g729, require
commercial licensing.
◦ Non Standard g726
▪ (asterisk:g726nonstandard)- If the peer negotiates G726-32 audio, use
AAL2 packing order instead of the RFC3551 packing order. This is required for Sipura
and GRandstream ATAs, among others. Contrary to the RFC3551 specification, the peer
should be negotiating AAL2-G726-32 instead.
◦ T38 Pass Through
▪ (asterisk:t38pt_udptl)- Enables T38 pass-through. This is used with SIP channels that
support sending and receiving of T38 FAX codecs to pass the call. Asterisk cannot
process the media.
Audio Codecs
Codecs
ulaw
g722
alaw
gsm
Ipc10
siren14
speex
siren7
adpcm
g723
slin
g726
g729
IIbc
g726aal2
© Copyright Schmooze Com, 2013
Non-Standard g726
Yes
No
T38 Pass-Through
Yes
No
Asterisk SIP Settings
User Guide
Video Codecs
• Video support can be enabled or disabled (default is disabled). When enabled, the following
settings are available.
◦ Codecs- As with audio codecs, check the desired codecs and drag to reorder. All others will
be disabled.
◦ Max Bit Rate- Maximum bit rate for video calls in kb/s.
Video Codecs
Video Support
Enabled
Disabled
h264
h263
h261
h263p
Max Bit Rate
384
kb/s
Media & RTP Settings
• Change the way media is handled during calls.
◦ Reinvite Behavior
▪ (asterisk:canreinvite)
• Yes- This uses standard reinvites.
• No- This equals never issue reinvites.
• Nonat- This is an additional option to allow media path redirection (reinvite), but
only when the peer that the media is being sent to is not behind NAT. The RTP core
can determine it based on the apparent IP address the media arrives from.
• Update- Use UPDATE for media path redirection instead of INVITE (yes = update +
nonat).
◦ RTP Timers
▪ (asterisk:rtptimeout)- Terminate call if “rtptimeout in seconds” of no RTP or RTCP
activity on the audio channel when the call is not on hold. This enables the ability to
hang-up a call in the case of the phone disappearing from the net, like a power outage or
someone tripping over a cable. Default is 30 sec.
▪ (asterisk:rtpholdtimeout)- Terminate call if “rtpholdtimeout in seconds” of no RTP or
RTCP activity on the audio channel when the call is on hold. Note that this value must be
greater then the “rtptimeout” value. Default is 300 sec.
▪ (asterisk:rtpkeepalive)- Send “keepalives” in the RTP stream to keep NAT open during
periods when no RTP stream may be flowing (like on hold). Default is 0.
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Asterisk SIP Settings
User Guide
MEDIA & RTP Settings
Reinvite Behavior
no
yes
RTP Timers
30
nonat
(rtptimeout) 300
update
(rtpholdtimeout) 0
(rtpkeepalive)
Notification & MWI
• Used to control polling frequency for MWI (Message Waiting Indication) and in controlling INUSE
information.
◦ MWI Polling Freq- Interval in seconds to check if the MWI state has changed and inform
peers. Default is 10 sec.
◦ Notify Ringing- Control whether the subscription already INUSE get sent ringing when
another call is sent. Useful when using BLF. Default is “Yes.”
◦ Notify Hold- Control whether the subscription INUSE gets sent ONHOLD when the call is
placed on hold. Useful when using BLF. Default is “Yes.”
Notification & MWI
MWI Polling Freq
10
Notify Ringing
Yes
No
Notify Hold
Yes
No
Registration Settings
• Control registration settings. Note that the defaults are generally sufficient.
◦ Registrations
▪ (asterisk:registertimeout)- Retry registration attempts every XX seconds until
successful or XX registration attempts have been made. Default is 20 sec.
▪ (asterisk:registerattempts)- Number of times to try and register before giving up. A
value of 0 means to keep trying forever. Normally you would leave this as the default of 0
so that Asterisk will continue trying to register until successful in the case of network or
gateway outages.
◦ Registration Times
▪ (asterisk:minexpiry)- Minimum length of registrations/subscriptions. Default is 60 sec.
▪ (asterisk:maxexpiry)- Maximum allowed time of registrations. Default is 3600 sec.
▪ (asterisk:defaultexpiry)- Default length of incoming and outgoing registrations. Default
is 120 sec.
© Copyright Schmooze Com, 2013
Asterisk SIP Settings
User Guide
Registration Settings
Registrations
20
(registertimeout) 0
Registration Times
60
(minexpiry)
(registerattempts)
3600 (maxexpiry)
120
(defaultexpiry)
Jitter Buffer Settings
• Jitter buffer temporarily stores arriving packets in order to minimize delay.
◦ Jitter Buffer
▪ (asterisk: jbenable)- Enable or disable the use of a jitter buffer on the receiving side of a
SIP channel. An enabled jiter buffer will only be used if the sending side can create a
jitter buffer and the receiving side cannot accept jitter. The SIP channel can accept jitter,
thus a jitter buffer on the receiving SIP side will be used only if it is forced and enabled.
For example, if receiving from a jittery channel to voicemail, the jitter buffer will be used if
enabled. However, it will not be used when sending to a SIP endpoint since they usually
have their own jitter buffers. See “jbforce” to force it’s use always.
◦ Force Jitter Buffer
▪ (asterisk:jbforce)- Forces the use of a jitter buffer on the receiving side of a SIP channel.
Normally the jitter buffer will not be used if receiving a jittery channel, rather the jitter will
be passed on to another SIP channel such as an endpoint that typically has a jitter buffer.
This will force the use of the jitter buffer before sending the stream on. This is generally
not recommended as it adds additional latency into the stream.
◦ Implementation
▪ (asterisk:jbimpl)- Jitter buffer implementation used on the receiving side of a SIP
channel. Two implementations are available:
• Fixed- This always equals “jbmaxsize.”
• Adaptive- This uses variable size (the new jb of IAX2).
◦ Jitter Buffer Logging
▪ (asterisk:jblog)- Enable or disable jitter buffer frame logging.
◦ Jitter Buffer Size
▪ (asterisk:jbmaxsize)- Max length of the jitter buffer in milliseconds. Default is 200.
▪ (asterisk:jbresyncthreshold)- Jump in the frame timestamps over which the jitter buffer
is resynchronized. Useful to improve the quality of voice with bug jumps in broken
timestamps. This anomaly is usually sent from exotic devices and programs.
© Copyright Schmooze Com, 2013
Asterisk SIP Settings
User Guide
Default is 1000 and it can be set to -1 to disable.
Jitter Buffer Settings
Jitter Buffer
Enabled
Force Jitter Buffer
Yes
Implementation
Fixed
Adaptive
Jitter Buffer Logging
Enable
Disable
Jitter Buffer Size
200
Disabled
No
(jbmaxsize)
1000
(jbresyncthreshold)
Advanced General Settings
• Change the way media is handled during calls.
◦ Language
▪ (asterisk:language)- The default language for a channel.
◦ Default Context
▪ (asterisk:context)- Default context for incoming calls if not specified. PBXact sets this to
“from-sip-external,” which is used in conjunction with anonymous SIP calls. If you
change this, it will effect that behavior. It is recommended that you leave this blank.
◦ Bind Address
▪ (asterisk:bindaddr)- The IP address to bind to and listen for calls on the bind port. If set
to 0.0.0.0, Asterisk will listen on all addresses. We recommend you leave this blank.
Asterisk 1.8 and above supports IPv6 and an address of “::” will listen on both IPv4/IPv6.
◦ Bind Port
▪ (asterisk:bindport)- This is the local incoming UDP port that Asterisk will bind to and
listen for SIP messages. The SIP standard is 5060 and in most cases you will use this
default. It is recommended that you leave this blank.
◦ Allow SIP Guests
▪ (asterisk:allowguest)- When set, Asterisk will allow guest SIP calls and send them to
the default SIP contact. Turning this off will keep anonymous SIP calls from entering the
system. However, the “Allow Anonymous SIP Calls” from the General Settings section will
not function. Allowing guest calls but rejecting the “Anonymous SIP Calls” in the General
Settings section will enable you to see the call attempts and debug incoming calls that
may be misconfigured and appearing as guests.
© Copyright Schmooze Com, 2013
Asterisk SIP Settings
User Guide
◦ SRV Lookup
▪ (asterisk:srvlookup)- Enable or disable. See current version of Asterisk for limitations
on SRV functionality.
◦ Call Events- Generate manager events when SIP performs events such as hold.
◦ Other SIP Settings- You may set any other SIP settings not present here that are allowed to
be configured in the General Settings section of “sip.conf.” There will be no error checking
against these settings, so check them carefully. They should be entered as “[setting]=[value].”
Click the “Add Field” button for additional lines. Blank lines will be deleted when you press
“Submit.”
Advanced General Settings
Language
Default Context
Bind Address
Bind Port
Allow SIP Guests
Yes
SRV Lookup
Enabled
Call Events
Yes
No
Disabled
No
Other SIP Settings
=
Add Field
• Don’t forget to press the “Submit Changes” button when done.
Submit Changes
Recap
• The Asterisk SIP Settings module affords easy access to the all-important SIP settings such
NAT, External IP, Local Network, Enabled Codecs and Codec order. It is important to
understand the ramifications of the various SIP settings as they greatly affect voice operations
and quality.
© Copyright Schmooze Com, 2013
Schmooze Com Inc.
(920) 886-8130
http://schmoozecom.com
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