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Table of Contents

What is the SBC capacity?

Sangoma provides two different tiers for its SBC. The Vega Session Controller and the NetBorder Session Controller. Both are based in the same software base, but Vega SBC is tailored to small densities (ie enterprise), from 1-250 concurrent calls. The NetBorder SBC it is aimed to big enterprises or ITSP/carriers, it goes all the way to 4,000 concurrent calls with hardware-assisted RTP/transcoding. In the near future we will at least double this capacity with more powerful DSPs and memory size.

The CPS (calls per second) measurement depends on many factors, including the hardware where you run it. Sangoma's SBC can run in standard Sangoma hardware appliances, custom hardware or even virtual machines. The carrier-level SBC appliance from Sangoma has been tested with 75 CPS with hardware transcoding involved.

What can it transcode?

Sangoma's NetBorder/Vega SBC does virtually all major codecs used in the industry, from narrow band (PCMU, G.729) to wide band codecs (ie G.722 and Siren/G.722.1 from Polycom)
The following is a list of supported codecs: G.711 (PCMU/PCMA), G.729, iLBC, G.722, G.722.1, GSM, G.723.1, G.726, AMR

The SBC is also capable of translating a variety of protocols, such as encrypted SIP TLS/SRTP traffic into non-encrypted UDP/TCP SIP traffic.

Where does it transcode?

Sangoma's NetBorder/Vega SBC is extremely flexible regarding transcoding. You can decide to do transcoding in hardware or software. You can also opt for bypassing media processing and allow the RTP flow directly between endpoints (this increases SBC overall capacity to handle more sessions easily). When doing hardware transcoding there is the option to do it built-in in the appliance DSPs, or with external DSPs (connected thru an ethernet network).

How does it ensure QoS (Quality of Service)?

There is several built-in mechanisms to protect QoS. You can specify CPU threshold limits to protect the quality of existing calls. Whenever the specified threshold is exceeded (ie, 80% CPU usage) the SBC will start refusing to accept new calls using a configurable SIP response code (ie 503 Service Unavailable), your equipment upstream can defer traffic to another SBC or gateway whenever it receives such code.

It is also possible to specify the ToS/DiffServ octet of SIP and RTP traffic to enforce QoS policies in the routing devices.

What kind of call routing does it do?

All the call routing is based on an XML scripting language, you can basically match SIP requests based on any field in the SIP packet (including source IP, SDP properties, codecs, headers etc) and route it to a defined SIP trunk/gateway or using the SBC built-in ENUM or LCR modules. You can also decide to reject the call or challenge the request yourself (this can also be done automatically by the SBC based on Call Admission Control rules).

How does it handle attacks?

There is multiple security mechanisms. The SBC comes with an IDS/IPS system (Intrusion Detection/Intrusion Prevention) system to block suspicious traffic. The definition of suspicious comes from a set of security rules/signatures of well-known VoIP attacks (there is rule sets for other protocols available as well, such as icmp, http).

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You can also detect malformed packets/traffic (someone trying sending garbage to see if it can crash your PBX or softw-switch), and the SBC can automatically block the offending IP address at the operating system level, where is extremely efficient to discard further packets from the offender.

How does it ensure reliability?

The SBC is capable of detecting SIP devices (ie gateways, proxies, soft-switches) that are down and re-route the traffic to alternate routes. This can be configured to be done automatically.

Does it provide ENUM support?

Yes, we support ENUM-based routing

Does it provide DTMF translation?

Yes we can translate from RFC2833 to inband

How well does it play with others?

We've done interoperability test with a number of PBX, phones and gateways. Some of them include:

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The list will keep growing in the coming weeks. If you do not see your vendor included in the list, we'll make it work for you. Our SBC is extremely flexible and we're confident that with the right configuration we can interop with any vendor.

Is there any limits for SIP trunks?

The number of SIP trunks you can create is only limited by the amount of memory (RAM) and hard-drive space available. In realistic situations you won't hit the limit ever, we've tested with 200 SIP trunks without problems (or even start to scratch any limit). Licensing is done only based on active calls, not on any other SIP dialog or request.

Is there any limits for Virtual IPs?

The number of Virtual IPs is unlimited.

How is SIP header manipulation done?

All header manipulation is done at the same time the routing is done in the XML script. Special variables define the meaning of different headers and parameters within those headers. This is an example of the INVITE URI modification in a SIP Refer request:

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Basically you match headers (ie ${sip_refer_to} is the variable where Refer-To header is populated by our SBC)when match against a desired regular expression,andthen replace either that same header or other headers by using the “export” or “set” application.

How can the RTP DTMF payload type be changed (ie from 96 to 101)?

You just set a variable during call routing before sending out the outgoing INVITE.

<action application="set" data="rfc2833_pt=96" />

If a SIP trunk is configured to use RFC2833 for DTMF but the remote end sends inband, can the SBC detect the tones?

Yes, the SBC can convert inband tones to RFC2833


Can your SBC can handle SIP and media/RTP on separate physical Ethernet lines/port.

Simple answerisYes.

Sangoma Carrier and Enterprise SBC perform RTP in hardware.
We can have two RTP operation modes in our SBC:  Exposed or Hidden
Exposed mode 
     Exposes RTP hardware IP addresses.
     RTP hardware communicates directly to remote agents via separate Ethernet port.
Hidden mode
     Hides RTP hardware IP address
     Single IP and Ethernetportis used for both Media and Signaling

Sangoma SBC also support VLAN's for both Signaling and Media/RTP.

 

What is the difference between calls and sessions?

This terminology may vary across vendors and even sometimes even within the same vendor some people may mistaken one for another. Be sure to clarify the meaning when comparing telecom equipment.

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If you acquire from our Sales organization an SBC with support for 250 sessions, you're getting an SBC with support for 250 sessions or calls (session and call is the same in this context). However, when you navigate through the WebUI (for example in the "Sessions" page) you will see 2 sessions per call (inbound/outbound legs) but you will be able to see up to 500 of these (twice as much as you have licensed).
 

How Does Call Forking work with the SBC?

With SIP Forking you receive one call, and the SBC as a result, forks into multiple calls (2 or more). Once one of the forked calls answers the first one in answering gets bridged, the other ones get cancelled. 

How to return multiple values from curl?

The response from curl is always stored in the variable $ {curl_response_data}, but you can return multiple values simply separate the values by commas, or any other character you want.

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Note as well how you cannot use the ampersand character in a curl request unless you escape it (because it's XML) using HTML entities (&amp;)

 

How can I replace my SSD drives in the even to a failure?

 

  1. Power off the Box

  2. Replace the FIRST SSD

  3. Power on the Box

  4. Navigate to WEBUI and run RAID1 synchronization

  5. Once synchronized, Power off the Box again

  6. Replace the SECOND SSD

  7. Power on the Box again

  8. Navigate to WEBUI and run RAID1 synchronization again

  9. Obtain the serial from the newly installed SSDs

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Here we have replaced the contents of the host (which was captured in variable $2) and only used $1 and $3 and inserted the desired hostname. The SBC will generate the new outbound SIP request with the new Diversion header value.

 

Can Virtual Machines Introduce Jitter?

Yes a virtual machine can introduce jitter into the audio stream. At the same rate other factors such as the network are the typical causes of jitter.

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The best defense against this is to ensure there is enough resources allocated to each VM. In addition to this ensure you never over allocate any resources on the host OS. As well it is always a good idea to keep your virtualization software up to date.

What is the cause of Error Parsing Kickstart Config

An error on line 36 of the kickstart file will generate a screen such as the following:

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