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Sip Trunking Providers

December 30, 2012, 10:17 pm
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This page is a list of SIP trunking providers around the world. Please keep this list in alphabetical order. SIP providers looking to add their services can do so in the list below.

Country specific pages:

VoIPInvite IncWholesale SIP termination service for dialer traffic. Call center using predictive dialer are welcome. 1000 of calls per second. no LRN, just npa nxx billing available. NPA NXX start from $0.0015. Canada, USA, international dialer termination available.

1comms VoIP provider for UK Businesses. Sip trunk services, UK DDI's and Asterisk Support for business users and resellers. White Label fully itemised per second billing.

1Pipe Telecom | OnePipe is a CLEC providing SIP Trunking, Hosted PBX, IP and other services, we provide services on our own network and gateways.

Alcazar Networks - Wholesale Services Over 3,100 DID rate centers. Per minute pricing as low as $0.0005/minute. Per channel pricing as low as $2.00/channel. DIDs as low as $0.10/each. A-Z termination. Over 1,200,000 numbers in DID inventory.

Amivox free your phone - Lower your communication cost VoIP provider for both consumers and businesses. Offer's free SIP account. Prepaid and very good rates for network termination with premium quality ( Amivox-Out) . Support for iPhone, Android and Blackberry. Shared balance for multiple users. Calling Amivox to Amivox is free - Sign up for free and try out the service.

Anveo offers phone numbers from over 48 countries with instant activation. Anveo's Voice 2.0 Communication and Collaboration Suite with powerful Visual Call Flow technology allows you to visually configure call handling and call termination options for your phone number. Anveo provides FREE SIP trunking and it is one of many termination options available.

BellVoz offers International and Domestic Long Distance Services with VoIP technology, helping business and consumers to reduce monthly telephony expenses.

Best VoIP USA BestVoIPUSA.com offers SIP trunking to private and commercial operators of Asterisk PBX switches. BestVoIPUSA.com also provides all of the well known features of SIP even if you are connecting directly to SIP compliant devices, handsets or servers.

Box Internet Services offers SIP trunking to private and commercial operators of Asterisk PBX switches. Boxis.net also provides all of the well known features of SIP even if you are connecting directly to SIP compliant devices or servers.

Brisnorth Communications Australia Brisnorth.com.au provides SIP trunks, VoIP and SIP Server hardware to Businesses Australia-wide. Carrier-Grade reliable SIP/VoIP services at very cost-effective rates. We can work with your current hardware/phones or upgrade you. We have Plans to suit all budgets and sizes of Business. ...
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VOIP Service Providers Business Asia

December 30, 2012, 10:38 pm
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This is a list of VOIP Service Providers who offer full service products primarily aimed at the small to medium sized business telephone market. Such companies typically support multi-line telephone systems, small PBX gateways and hosted VoIP (as an alternative to Centrex service). See also:

Page Contents



Service providers operating in more then one country are listed under each country. "Operating in a country" means a provider that has billing and support staff located in the country, and offers service subject to the regulations and consumer protection laws of that country.

Marketing is NOT ALLOWED on this page. Please describe services in neutral language and normal fonts. Don't bother listing prices--unless you really plan to return and edit them as things change. If you want to add your company, please read the Posting Guidelines for Promoting Products and Services. When you add your entry to this page, please make sure your entry is in alphabetical order in relationship to other vendors listed in the same section. Failure to follow these guidelines will result in deletion!


Please include relevant information like SIP or IAX handoff, how outgoing CLID is set, whether a given account may originate multiple calls at once, etc.

If you like this page, please link to it, so Google and other search engines will consider it more important.

Users: Please feel free to REMOVE any listing that does not meet the stated goals of this page. ...
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FreePBX CallerID Lookup

December 31, 2012, 12:42 am
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Description:


Add new number & callerid name on "Asterisk Phonebook" from FreePBX GUI. FreePBX uses asteirsk's astdb/cidname as the database.(do a "database show cidname" from Asterisk CLI)

1.png


Add a New Entry in CallerID Lookup Sources

2.png


From Inbound Routes select the "CID Lookup Source" as the newly added Source.

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xCALLY

January 1, 2013, 1:27 am
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xCALLY is a professional Contact Center Suite optimized for Asterisk.

xcally3logosmall.png

Developer: Xenialab
Released: 2010-2011
Official Web Site: www.xcally.com

Features

- Inbound MultiCampaigns, MultiSkills and MultiChannels
- Outbound Automatic Dialer
- Blended features

Description

xCALLY is a modular architecture developed using hybrid .Net and Linux components.
The main modules are:
1. xCALLY Telephone Bar: a phone bar designed for the customer care agents, developed to manage the agent status with Login/Logout, Pause, Backoffice, Call features, Call Pop-ups, CTI Computer Telephony Integration for external CRM, ERP and general third party applications and API.

xCallyBar.png

2. xCALLY Core module: a proxy server to connect multiple telephone bars along with the connection to Asterisk for call routing
3. xCALLY Reports: a Web Interface to analyze offline and realtime optimized reports for contact centers

xcallyreports.png

4. xCALLY Manager GUI: a Web Interface to configure the multi-campaigns, multi-slkills based-routing, time intervals, DID, Centralized Address Books, Call Lists, Sounds etc...

managergui.png

5. xCALLY Automatic Outbound: a module to manage extension lists and outbound algorithm to perfom automatic outbound distribution calls

outbound.png


Inbound MultiChannels

xCALLY can dispatch inbound services based on:
- VoIP or Traditional DID: Toll-Free numbers or normal extensions routed through SIP Trunks or traditional PSTN lines
DID.png


- Web Calls using the Hand4Shake Web Communication widgets
click2call.png

- xCALLYMPP: Messaging Precense Panel, optimized for Facebook, Google+ and XMPP services

xcallymppFCB1.png
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Phone Numbers

January 1, 2013, 8:53 am
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You can call the following VoIP numbers for free. From a softphone, just copy&paste the full SIP address (sip:user@domain.com).
From a hardware SIP adapter, copy & paste the full SIP address into your speed dial web page. All of these numbers worked in July 2011.

Dialing through PSTN



Dialing through SIP URI (all destinations listed working as of Jan 1, 2013)


  • Free 411 sip:411@ideasip.com +1-800-FREE411 US free directory assistance


  • Mouselike.org (UK) Echo and audio quality tests sip:904@mouselike.org or PSTN:+441483604781 - Allows test / connection from anywhere.

  • Telephreak sip:telephreak@voip.telephreak.org - The Telephreak's Free Voicemail/Conference system


  • UCLA sip:13108254321@ucla.edu (no G.729)

  • QXIP Music on Hold sip:9999@qxip.net (Curtesy of QUASARMUSIQ)

  • QXIP Streaming sip:9901@qxip.net (Hypemachine RSS Feeds)

  • QXIP Streaming sip:9902@qxip.net (Misc. ...
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Teamspeak

January 1, 2013, 9:14 am
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TeamSpeak

http://www.goteamspeak.com

TeamSpeak is an application which allows its users to talk to each other over the internet and basically was designed to run in the background of online games. The voice quality and hereby the bandwidth usage is configurable and can be low for modem users or as good as normal phone calls for better connections. Still the client's maximum bandwidth won't exceed 25.9 KBit/s (=3.3 KB/s) upstream.
Also TeamSpeak uses as little latency as possible, which makes TeamSpeak one of the best online gaming communication programs.

But one of TeamSpeak's largest advantages is: TeamSpeak is a cross-platform voice communication tool. This means that client and server are available for Windows as well as for Linux.

Additionally we have to say that TeamSpeak is a contribution to the gaming community and thus it is completely free for non-commercial users.

See also: Ventrilo

See also: C3


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Ventrilo

January 1, 2013, 4:18 pm
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Ventrilo

Ventrilo Site

Ventrilo is a very simple and easy to use VoIP program designed for multiplayer gaming. It is functionally similar to TeamSpeak, but has a much simpler interface and feature set.

Clients are available for Win32, with Linux/i386 and Mac OSX in development.
Servers are available for Win32, Linux/i386, Solaris/i386, Solaris/SPARC 64bit, FreeBSD/i386, NetBSD/i386 and Mac OSX.
Both client and server are closed-source.

Client software is free. However recent licensing changes with v2.2.x have made the free server crippleware, only runs one instance per machine on the default port only with up to 8 users only.
The full server version is only available to those reselling Ventrilo hosting, if you aren't reselling Ventrilo they will refuse to sell you a server.
Because of this licensing system I recommend TeamSpeak instead.


See also: TeamSpeak

See also: C3


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C3

January 1, 2013, 4:18 pm
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C3 is a hosted and free voice application geared towards online gamers (TeamSpeak, Ventrilo style).

http://www.downloadc3.com

Platforms supported: Windows and Mac OS X

- Hosted and free to use (unlimited channel size)
- Multi-channel (adjustable listen/speak groups of channels/sub-channels)
- Free PSTN calls into channels
- Moderator controls
- Password protected channels

- Windows and Mac only


See Also:

Ventrilo

Teamspeak


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Thomson ST2030

January 1, 2013, 5:56 pm
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Technicolor ST2030 an IP Phone for Enterprise Market

Phone presentation here!
Image
Image

Page Contents

Latest SIP firmware release version x.78 (x=1 for normal, x=8 for new "secure" version). Downloadable at http://www.technicolorbroadbandpartner.com. Direct Link (might not work if out-dated)

Features available in both SIP and MGCP versions

The contemporarily-designed ST2030 will be enjoyed by your end users.
It offers all the features of a professional phone:
  • Full graphic display 128x64
  • 3 softkeys for intuitive navigation
  • 4-way conferencing
  • Powerful and High Quality handsfree mode
  • Call forwarding, call parking and transfers
  • 10 flashing line keys for speed dial and monitoring extensions. ...
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ZONE Limited

January 2, 2013, 12:21 am
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just ZONE-smaller2.gif
http://www.zonetel.com


Established in 2000, ZONE Limited is a licensed telecom service and solution provider in Hong Kong. ZONE specializes in VOIP/SIP/Asterisk PBX/Call Center/IVR and have a number of successful implementations in Asian regions. ZONE endeavours to provide customers with best products and professional services.

NEW

Elastix minUCS box

The Elastix miniUCS series are compact all-in-one IP PBX for SMB featuring:

Fxo or T1/E1 connectivity
Small size — less space, less power consumption
Same PBX functionality as the Elastix

For more details, please visit here

Services

DID numbers/Hong Kong Virtual Phone Number

ZONE is a licensed telecom operator offering HK DID numbers (HK virtual phone numbers) to your compatible SIP end points like Asterisk, Elastix, FreePBX, Microsoft Lync and even your Android or iPhone. You can get HK presence without all the expensive overhead. The DID comes with optional outbound dialing, voice recording, voice mail, call forward and concurrent call support.

Many call centers and virtual offices are using our HK DID services.

In addition to HK DID, we provide Singapore and China DID for you to develop the Asia markets.


IP-PBX

We carry IP-PBX products from selected brands which we believe to represent the best mix to serve SME/SOHO segments. We provide on-site installation, integration, customization and training so that customers can make full use of the IP technology.

Asterisk consultation and turnkey solutions

Asterisk is known to be the leading open-source PBX attributed to its powerful features and deployment flexibility. We have intensive exposure to Asterisk and could help you to implement various business applications such as calling card, IVR, voice recording, fax2email, auto-attendant, dialers, etc.

Cloud/Hosted PBX and IVR service

Looking for a hosted PBX service? We provide elastix/asterisk PBX hosting environment with DID SIP trunks and support of remote extensions. Pay-as-go-you.
Or need a hosted IVR service to integrate with your business applications ? We are able to deliver advanced features like call queues with customizable greetings, operator-assisted key, fax-on-demand, time-dependent IVR options, database integration and many.

Please visit Cloud Plans for price quotes.

SMS service with international coverage

It is convenient to broadcast and schedule SMS on our web site. In addition, we act as a programmable SMS gateway supporting the standard smpp and http interface.


A-Z SIP termination

ZONE has been operating IDD business in HK for more than 10 years. We partner with many international carriers to provide a reliable and quality voice service. We have also extended the IDD service via VOIP/SIP so that you could enjoy our attractive A-Z rates from anywhere in the globe.

Web callback

Launching toll-free access to your customer service center becomes easy and cost-effective with our web callback. It just involves a few lines of codes in your web server to enable the toll-free channel with every usage statistics. ...
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VOIP sites

January 3, 2013, 7:03 pm
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Links to other Non-Commercial VoIP sites

This page is intended for informative and/or community related websites only. Postings of websites from manufacturers, VoIP networks, consultancy, free calling, calling cards and other commercial websites will be removed. See also Asterisk news and blogs. Links out of URL order are also subject to deletion. (hint, make your display tag the same as your domain name)

A

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Asterisk cmd jack

January 3, 2013, 9:47 pm
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app_jack

Synopsis

Jack Audio Connection Kit

Syntax

JACK([options])

Common options:

s(<name>): Connect to the specified jack server name
i(<name>): Connect the output port that gets created to the specified jack input port
o(<name>): Connect the input port that gets created to the specified jack output port
n: Do not automatically start the JACK server if it is not already running
c(<name>): By default, Asterisk will use the channel name for the jack client name.
Use this option to specify a custom client name.

Asterisk Version

Introduced in 1.6.x?

Description

This is an application to connect an Asterisk channel to an input and output jack port so that the audio can be processed through another application, or to play audio from another application.

When the JACK() application is executed in the Asterisk dialplan, two JACK ports get created. There is an input and output port that acts as the endpoint of a phone call. The audio from the channel goes out of the output port that gets created. Whatever audio that comes in on the input port is what gets sent back to the caller. This allows for some advanced voice applications that interact with the audio of the call and run as a separate application.

When you use the JACK application, you get:

Incoming call audio -> Jack Application
Outgoing call audio <- Jack Application

JACK_HOOK() function: This interface is a little bit more complex, but it is the much more interesting one, in my opinion. The JACK_HOOK function creates an audiohook and attaches it to the channel. In this case, instead of the JACK interface being the endpoint of the phone call, it is simply hooked in to the audio path for a phone call to something else. Audio that comes from a caller gets sent out the JACK output port, and whatever audio that comes back in on the input port gets sent on as the caller’s audio. This allows for cool applications that do analysis and manipulation of the audio in a phone call. One example is that I can now write custom vocoders in Pd.

JACK_HOOK does the following:

Incoming call audio -> audio in to jack, audio out from jack -> current Asterisk application
Outgoing call audio <- current Asterisk application

Details


Have you ever wanted to take the audio output of one piece of software and send it to another? How about taking the output of that same program and send it to two others, then record the result in the first program? If so, JACK may be what you’ve been looking for.
JACK is a low-latency audio server, written for POSIX conformant operating systems such as GNU/Linux and Apple’s OS X. It can connect a number of different applications to an audio device, as well as allowing them to share audio between themselves. Its clients can run in their own processes (ie. as normal applications), or can they can run within the JACK server (ie. as a “plugin”).


Examples

exten => 1234,1,Answer
exten => 1234,n,JACK()

exten => _1NXXNXXXXXX,1,Answer
exten => _1NXXNXXXXXX,n,Set(JACK_HOOK(manipulate)=on)
exten => _1NXXNXXXXXX,n,Dial(IAX2/myprovider/${EXTEN})

  • CLI> core set chanvar SIP/poly1-ab23jadf234 JACK_HOOK(manipulate) on

Note: the command above does not connect A and B legs, only opens jack ports. ...
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Asterisk func shared

January 3, 2013, 10:16 pm
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Synopsis

Gets or sets the shared variable specified
Introduced with Asterisk 1.6.1

Description


SHARED(<varname>[,<channel>])

Implements a shared variable area, in which you may share variables between channels. If channel is unspecified, defaults to the current channel. Note that the channel name may be the complete name (i.e. SIP/12-abcd1234) or the prefix only (i.e. SIP/12).

The variables used in this space are separate from the general namespace of the channel and thus ${SHARED(foo)} and ${foo} represent two completely different variables, despite sharing the same name.

Finally, please realize that there is an inherent race between channels operating at the same time, fiddling with each others' internal variables, which is why this special variable namespace exists; it is to remind you that variables in the SHARED namespace may change at any time, without warning. You should therefore take special care to ensure that when using the SHARED namespace, you retrieve the variable and store it in a regular channel variable before using it in a set of calculations (or you might be surprised by the result).

Usage

On each channel, there is a space accessible for other channels to write:

Set(SHARED(foo,SIP/123)=456)

or retrieve:

${SHARED(foo,SIP/123)}

The primary reason for having this space is writing out to another channel, since you can already import variables (and functions) from another channel, with the IMPORT function:

${IMPORT(SIP/123,CALLERID(name))}

Just remember that this is a special variable space and not the main variable space, so that other channels cannot mess with your execution except when you explicitly want them to be able to do so.

See also


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Asterisk cmd Set

January 3, 2013, 10:16 pm
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Synopsis

Sets variable to value

Version differences:
This command is not available in Asterisk 1.0.9. Use SetVar instead.
As of v1.2SetVar is deprecated and we are back to Set.
As of v1.4 the use of Set() to set multiple variables at once and the g flag have both been deprecated. Please use multiple Set() calls and the GLOBAL() dialplan function instead.
As of v1.6 only if you have a corresponding "compat" setting, then Set() does not strip surrounding quotes from the right hand side as it did previously.


Description


Set(variablename=value|options]) (starting with Asterisk 1.4)
Set(variablename=value[|variable2=value2][|options]) (up to and including Asterisk 1.2)

This application can be used to set the value of channel variables or dialplan functions. It will accept up to 24 name/value pairs upto Asterisk 1.2, but only one name/value pair in Asterisk 1.4 or later.
When setting variables, if the variable name is prefixed with _ the variable will be inherited into channels created from the current channel. If the variable name is prefixed with __ the variable will be inherited into channels created from the current channel and all children channels.
Next to the Set() command there is also the SET function available.

Options for Asterisk 1.2 (and possible earlier versions)

  • g: set a global variable (valid in the entire dialplan, not just the channel)


As of Asterisk 1.4 the correct syntax to set a global variable is like this

exten => 100,1,Set(GLOBAL(FOO)=456)

extensions.conf:
; If clearglobalvars is not set, then global variables will persist
; through reloads, and even if deleted from the extensions.conf or
; one if its included files, will remain set to the previous value.
;
clearglobalvars=no

Asterisk 1.6

Note that Set() changes behaviour in Asterisk 1.6 which can be controlled via asterisk.conf:

[compat]
app_set=1.6

If (and only if), in /etc/asterisk/asterisk.conf, you have a [compat] category, and you have app_set = 1.6 under that, then the behavior of this app changes, and does not strip surrounding quotes from the right hand side as it did previously.

Example

Set(numTries=4)
Set(CALLERID(number)=000000)
Set(CALLERID(name)="The Name")
Set(NIGHTMODE=1,g) ; set a global variable

To increment a variable, you can use:

Set(total=$[${total} + 1])

To set inherited variable:

Set(_CALLID=${UNIQUEID})
Set(__CALLID=${UNIQUEID})

NOTES:
  • Variable names are not case sensitive.
  • Each channel gets its own variable space. There is no chance of collisions between different calls, and the variable is automatically trashed when the channel is hangup.
  • Make sure you do not put spaces around the equals sign in the assignment. Set(numTries = 4),with a space on either side of the "=", will set numtries to "". ...
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voip-info.org

January 3, 2013, 10:50 pm
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Welcome to the VOIP Wiki - a reference guide to all things VOIP.


This Wiki covers everything related to VOIP, software, hardware, VoIP service providers, reviews, configurations, standards, tips and tricks and everything else related to voice over IP networks, IP telephony and Internet Telephony.

Your contributions are welcome, please read the How to add information to this wiki page and the Posting Guidelinesbefore you post.

Update: We have added theFacebook Like and Google +1 button to the top right corner of all pages on Voip-Info.org. Please help recommend the wiki by clicking on them. We also now have Google+ page here and a Facebook page here. Visit them and add us. Thanks!


NEWS


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Ecessa

January 4, 2013, 4:38 am
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Ecessa-Logo.png




Ever since our founding in 1968, we have been at the forefront of the constant stream of innovations that define the telecommunications industry. Back then, essentially the dawn of the modern IT industry, we were in the business of designing, manufacturing and distributing telecommunications for local area networks.

Jump ahead to 1989 when we met the voracious need for greater network reach with the development of several high-speed digital access communication products that linked facilities through wide area networks.

In 2000, the company refocused its products on appliance-based WAN link failover and load balancing technologies to give these expanded networks greater reliability, bandwidth and redundancy. We did this initially with PowerLink™, which bundles multiple WAN lines (T1, xDSL, Cable, ISDN, Wireless, Satellite, etc.) into a greater pool of available bandwidth while providing business continuity through ISP failover and redundancy.

Since then we’ve added ShieldLink™, our family of advanced, yet highly affordable and secure WAN Link Controllers that incorporate all the capabilities of PowerLink™, while bringing network security to the next level by incorporating a built-in firewall and VPN Gateway. And then the introduction of ClariLink™, our newest product addition, providing real-time failover for VOIP reliability over the Internet, and optimizing performance by managing traffic among multiple Internet links to avoid jitter, latency and congestion.

Today, we continue this innovation tradition, with the introduction of our WaaS service....

WaaS Service Highlights and Features:


WaaS offers many great benefits for small and medium-sized businesses that rely on their Internet connectivity and cannot afford a moment of Internet downtime but don’t have thousands of dollars to invest in expensive hardware.

Here is a list of a variety of features that are sold a la carte within our WaaS offering so that businesses only pay for what they need/want:

  • VoIP Failover and Monitoring
  • Multi-ISP WAN Failover
  • Intelligent Inbound and Outbound Load Balancing
  • Proactive monitoring and reporting of their WAN network
  • Network Traffic and Application Visibility
  • Traffic Shaping and Application Prioritization
  • Firewall and VPN
  • WAN Virtualization and Line Bonding between sites

One of the most innovative parts of our WaaS offering is our VoIP Failover and Monitoring feature. When activated, this feature turns on our micro-appliance’s ability to monitor all VoIP calls (incoming and outgoing) in real time. Then in the event of a WAN failure or call degradation (i.e. high latency, excessive jitter, packet loss, etc…) our service will automatically move active VoIP calls to other up and running or better performing WAN links, without the call dropping or the user even noticing.

In effect, this means that business VoIP users and providers never have to worry about having another dropped or low quality VoIP call again. Ecessa is the only service/product offering this fantastic feature (patent pending).

Our affordable service is paid monthly so there are no major upfront costs (which is perfect for small and mid-sized companies) and virtually guarantees businesses a ‘worry proof’ Internet and VoIP experience. Click below for more detailed information:

WAN Optimization as a Service


Company News


2012.12.14 - Ecessa's 'WaaS' Service Receives Product of the Year Award
2012.10.04 -
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VOIP Service Providers Business Europe

January 4, 2013, 4:53 am
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This is a list of VOIP Service Providers who offer full service products primarily aimed at the small to medium sized business telephone market. Such companies typically support multi-line telephone systems, small PBX gateways and hosted VoIP (as an alternative to Centrex service). See also:


Service providers operating in more then one country are listed under each country. "Operating in a country" means a provider that has billing and support staff located in the country, and offers service subject to the regulations and consumer protection laws of that country.

Please describe services in neutral labanguage and normal fonts. Don't bother listing prices--unless you really plan to return and edit them as things change. If you want to add your company, please read the Posting Guidelines for Promoting Products and Services. When you add your entry to this page, please make sure your entry is in alphabetical order in relationship to other vendors listed in the same section. Failure to follow these guidelines will result in deletion!


Please include relevant information like SIP or IAX handoff, how outgoing CLID is set, whether a given account may originate multiple calls at once, etc.

Users: Please feel free to REMOVE any listing that does not meet the stated goals of this page. ...
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VOIP Service Providers Residential

January 4, 2013, 4:55 am
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VOIP Phones

January 4, 2013, 4:59 am
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This page is for listing brief details of VoIP Phones including details, where to buy, specifications, and any other relevant VoIP Phone information. Please read the Posting Guidelines for Promoting Products and Services before adding to it.

Hard Phones

Standalone Ethernet Hard Phones (voice only)
An Ethernet hard phone is a self contained IP telephone that looks just like a conventional phone but instead of a conventional phone jack, it has an Ethernet port through which it communicates directly with a VoIP server, VoIP gateway or another VoIP phone. Since a broadband hard phone communicates directly with a VoIP server, VoIP gateway or another VoIP phone it does not require any personal computer nor any software running on a personal computer to make or receive VoIP phone calls. It can be used independently, all that is required is an internet connection. While PC based software solutions are cheaper, a hard phone is the best solution for IP telephony.

Page Contents


General

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Asterisk RTCP

January 6, 2013, 2:37 am
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Before patch 10590 (which was applied to Asterisk 1.6) the RTCP data presented in Asterisk 1.4 were basically useless: For example packet loss values could be 10x higher than the number of total received packets, and only one of the two calls legs was being watched.

See Patch 10590 (updated and committed in June 2008)
"rtpqos-14-r119891.diff" is meant for Asterisk 1.4 and does also apply to Asterisk 1.4.30 after minimal manual intervention. Since Asterisk 1.4 does not receive new features this has not made it into subsequent 1.4 realeases.
Note: Packet2Packet bridging (aka p2p) will completely mess up the RTCP stats with this patch with up to wrongly reported 100% packet loss! So either make sure that p2p bridging does not occur (by using a force-enabled jitter buffer or by making sure transcoding is done), or disable the p2p code in rtp.c (which is really simple to do).

In Asterisk 1.6 the RTCP data are now provided with the help of the CHANNEL function.

Useful CLI commands

rtcp stats
sip show channelstats (Asterisk 1.6)
rtcp debug <ip>


Quality rules

  • Packet loss < 1% (preferably 0.5%); with a packet loss of > 3% call quality will degrade audibly, at > 6% quality turns bad, and above 10% a call becomes entirely unacceptable.
  • Delay/Latency < 150 ms (one way); to much delay can aslo affect/kill echo cancellation (think satellite link with 500-700 ms delay)
  • RTT/Round Trip Time < 200-250 ms (RTT = delays of both directions added)
  • Jitter < 30 ms. The meaning of the jitter value depends greatly on the jitter buffers involved. Jitter affects the packet loss
  • "Bursts" of consecutively lost packets are bad bad bad (aka "drop out")
  • codecs: Some codecs (g729, iLBC) are better at concealing a lost packets than others (g711), keyword "PLC": Packet Loss Concealment
  • TOS/QoS/DiffServ settings can help to prioritize VoIP traffic on that part of the network you have control over


Questions

  • Is it mandatory to have a proper NTP time setup to issue (and correctly honor received) RTCP reports? A: No, but it helps.
  • What are the dimensions of the reported values - typically they are in seconds, however "reported jitter" appears to be shown in milliseconds?
  • Is there a way to match a SSRC to a SIP UA, a call leg or call's unique ID? No, not easily, the SSRCs change often.


Abbreviations

RTCP: RTP Control protocol
SSRC: Synchronization Source
SR: Sender Report (issued by sites that also sent data in the past interval)
RR: Receiver Report (issued by sites that only received data)
RTT: Round Trip Time (how long does it take to the other side and then back to us = both delays/latencies added)
QOS: Quality of Service
rx: Receive
tx: Transmit (send)
Transit: Relative transit time for previous packet
DLSR (and LSR) allow a sender to calculate the round trip time of RTCP reports. (minus the internal delay on the remote side)
avg: average
stdev: standard deviation (statistics)

Examples

Example 1: CLI statistics of an unpatched Asterisk 1.4:

Note: There are two call legs with each a "sender" and a "receiver"

  • Our Receiver:
SSRC: 1562797448
Received packets: 851
Lost packets: 0
Jitter: 0.0032
Transit: -0.0041
RR-count: 0

  • Our Sender:
SSRC: 1172311862
Sent packets: 798
Lost packets: 0
Jitter: 0
SR-count: 3
RTT: 0.026000

Example 2: CLI output of "rtcp debug ip aa.bbb.cc. ...

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