In this section:
SBC SWe Lite Call Performance
The table below includes the maximum call performance numbers for the listed call flow.
Note: For details on calculating DSP Requirements and determination of VM attributes to host the SBC SWe Lite, refer to Calculating DSP Requirements for SBC SWe Lite.
Call Performance - KVM, VMware, Microsoft Hyper-V On-Premises Deployments 1 Maximum number of concurrent sessions. The number assumes that calls are made using RTP/SRTP Proxy mode, or a mix of RTP/SRTP Proxy, media manipulation and video calls. 2 Maximum number of concurrent sessions with virtual DSP intervention. See Transcoding Capacity below for details. 3 Maximum number of concurrent audio/video sessions. The total system capacity is affected if A/V calls are introduced into the call mix. Maximum number of calls is reduced by the number of video streams used. For example, 1 vCPU instance processing 25 A/V calls has a total capacity of: 300 (max number of calls) - 25 (calls processed with 1 vCPU instance) = 275 calls. 4 Maximum number of concurrent sessions (when virtual DSP intervention is applied to 450 sessions) is 1000. 5 Maximum number of concurrent sessions (when virtual DSP intervention is applied to 600 sessions) is 600. 6 Maximum number of proxy media mode concurrent sessions is reduced by a count equivalent to the active number of concurrent RTP media manipulation sessions. Refer to note 1. 7 Supported by *-SG/*-SGX/*-SP licenses. 8 Supported by *-SGX/*-SP licenses. 9 Supported only by *-SP licenses. 10 Supported by -SGX/-SG/-SP. See "SIP Signaling & RTP Direct Media/RTP Proxy Sessions without Encryption Services" in Working with Licenses. Modified: for 8.1.5 Call Performance - Microsoft Azure Cloud 1 Maximum number of concurrent sessions. The number assumes that calls are made using RTP/SRTP Proxy mode, or a mix of RTP/SRTP Proxy, media manipulation and video calls. 2 Maximum number of concurrent sessions with virtual DSP intervention. See Transcoding Capacity below for details. 3 Maximum number of proxy media mode concurrent sessions is reduced by a count equivalent to the active number of concurrent RTP media manipulation sessions. Refer to note 1. 4Supported by -SG-CLOUD/-SGX-CLOUD/-SP-CLOUD licenses. 5 Supported by -SGX-CLOUD/-SP-CLOUD licenses. 6 Supported by -SP-CLOUD license. The number of RTP Port Pairs must be configured slightly larger than the actual number of ports required to support the projected number of calls. We recommend you over-allocate the number of port pairs by approximately 25 - 30% above the number of calls you want to support. For details, see Configuring the Media System.
SWe Lite Virtual Machine Resources, applicable to all supported hypervisors (KVM, VMware® Microsoft Hyper-V) Maximum SIP
with corresponding
RTP Media Sessions1 SIP Signaling Session Limits RTP Media Session Limits Maximum Call Rate Setup (CPS) Media Manipulation Mode2 (Requires Virtual DSP Intervention) Proxy Media Mode (No Virtual DSP Intervention) Audio/Video Streams 3 vCPU # GB RAM Maximum TCP/TLS-based SIP↔SIP Signaling Sessions7 Maximum SIP Registrations (60 minute refresh rate) No transcode, with in-band services scenario8 Default scenario: G.711/G.729ab RTP ↔ G.729ab/G.711SRTP, with in-band services9 Encryption services: G.711 RTP ↔ G.711 SRTP8 No encryption services: RTP ↔ RTP/SRTP ↔ SRTP10 1 1 GiB 100 300 1000 100 100 300 6 300 6 25 10 2 1.5 GiB 1000 1000 1000 200 200 1000 6 1000 6 50 10 4 2.5 GiB 1000 1000 4 /600 5 5000 450 4 /600 5 450 4 /600 5 1000 6 1000 6 100 10 10 2.5 GiB 1000 1200 5000 1200 1200 1200 1200 100 10
Azure Virtual Machine (VM) Resources Maximum SIP
with corresponding
RTP Media Sessions1 SIP Signaling Session Limits RTP Media Session Limits Maximum Call Rate Setup (CPS) Media Manipulation Mode2 (Requires Virtual DSP Intervention) Proxy Media Mode (No Virtual DSP Intervention) VM Instance vCPU Maximum TCP/TLS-based SIP↔SIP Signaling Sessions4 Maximum SIP Registrations (60 minute refresh rate) No transcode, with in-band services scenario5 Default scenario: G.711/G.729ab RTP ↔ G.729ab/G.711SRTP, with in-band services6 Encryption services: G.711 RTP ↔ G.711 SRTP5 No encryption services: RTP ↔ RTP/SRTP ↔ SRTP4 B1ms 1 10 10 100 10 10 10 3 10 3 10 B2s 2 100 100 500 30 30 100 3 1003 10 DS1_v2 1 300 300 1000 100 100 300 3 300 3 10 DS3_v2 4 1000 1000 5000 400 400 500 3 1000 3 10
Number of RTP Port Pairs must be increased above maximum call capacity
Call Capacity Limitations
Transcoding Capacity
The table below indicates the supported codecs and the maximum number of concurrent transcoded calls for specific codec combinations and system size. For the supported code list, refer to Protocols and Functions Supported.
Transcoding Capacity - KVM, VMware, Microsoft Hyper-V On-premises Deployments (1vCPU, 2vCPU, 4vCPU)
Transcoding Capacity - KVM, VMware, Microsoft Hyper-V On-premises Deployments (1, 2, 4 vCPU)
| Transcoding Scenario | Virtual Machine vCPU Count | |||
| CODEC 1 | CODEC 2 | 1 vCPU | 2 vCPU | 4 vCPU |
|---|---|---|---|---|
| G.711A-law or G.711u-law | G.711A-law or G.711u-law | 100 | 200 | 600 |
| G.711A-law or G.711u-law | G.723 | 80 | 160 | 480 |
| G.711A-law or G.711u-law | G.726 or G.729 | 100 | 200 | 600 |
| G.711A-law or G.711u-law | AMR WB | 38 | 76 | 225 |
| G.711A-law or G.711u-law | Opus | 24 | 54 | 165 |
| G.711A-law or G.711u-law | T.38 | 50 | 100 | 300 |
Transcoding Capacity - KVM, VMware, Microsoft Hyper-V On-premises Deployments (10 vCPU)
Transcoding Capacity - KVM, VMware, Microsoft Hyper-V On-premises Deployments (10 vCPU)
| CODEC 1 | CODEC 2 | 10 vCPU |
|---|---|---|
| G.711A-law or G.711u-law | G.711A-law or G.711u-law | 1200 |
| G.711A-law or G.711u-law | G.729 | 1200 |
| Opus | G.711A-law/mu-law | 360 |
Transcoding Capacity - Microsoft Azure Cloud Deployments
Transcoding Capacity - Microsoft Azure Cloud Deployments
| Transcoding Scenario | Microsoft Azure VM Instance | ||||
| CODEC 1 | CODEC 2 | B1MS VM | B2S VM | DS1_v2 | DS3_v2 |
|---|---|---|---|---|---|
| G.711A-law or G.711u-law | G.711A-law or G.711u-law | 10 | 30 | 100 | 400 |
| G.711A-law or G.711u-law | G.726 or G.729 | 10 | 30 | 100 | 400 |
SILK Capacity
The SBC Edge supports the SILK audio codec. Skype designates SILK as an internet wideband audio codec for use in VoIP. SILK operates at two different sampling rates: 8000 Hz narrowband and 16,000 Hz wideband (see the SILK Bandwith Options table). These rates allow for the capture of higher frequencies, which provide fuller sound, while also allowing interoperability with the Public Switched Telephone Network (PSTN). SILK has Low Bit Rate Redundancy (LBRR), also called Forward Error Correction (FEC), which protects the SBC Edge against packet loss. The network bit rate of SILK is adaptive within the range that the following table specifies. The SBC Edge defines and modifies the average network bit rate in real-time, while the actual bit rate depends on the input signal and change over time. The bit rate can dynamically change within that range. Since all other parameters are equal, the higher bit rates result in higher audio quality. SILK Bandwidth Options Narrowband Wideband
Audio Bandwidth Frequency (Hz) Bit Rate (KBPS) Description 8000 6 - 20 The SBC Edge only uses the narrowband mode either to interface to PSTN networks, or on low-end devices that support 8000 Hz or less sampling frequency.16,000 8 - 30 The SBC Edge uses the wideband mode for all IP platforms that support 16,000 Hz or less sampling frequency.
The following tables outline the SILK performance and capacity. The following table outlines the SILK performance numbers for Microsoft Azure Cloud deployments. SILK Performance for Microsoft Azure Cloud Deployments The following table outlines the SILK performance numbers for On-premises deployments. SILK Performance for On-premises Deployments 95 SILK Performance for Microsoft Azure Cloud Deployments
Transcoding Scenario Microsoft Azure VM Size B1ms B2s DS1_v2 DS3_v2 SILKNB SRTP -> G711U RTP 10 30 95 400 SILKNB SRTP -> G729A RTP 10 30 50 300 SILKNB SRTP -> SILKWB RTP 10 30 35 200 SILK Performance for On-premises Deployments
Codec Virtual Machine vCPU Count 1 vCPU 2 vCPU 4 vCPU SILK-WB-G711U 55 135 340 SILK-WB-G729 35 95 230 SILK-WB-SILK-NB 35 88 215 SILK-NB-G711U 95 200 595 SILK-NB-G729 50 130 325 G711U-SILK-NB 200 595 G729-SILK-NB 50 130 325 SILK-NB-WB 35 88 215
SBC SWe Lite Single Instance Capacities
The SBC SWe Lite supports the following maximum configuration.
SBC SWe Lite Maximum Configuration Values
Feature | Maximum supported |
|---|---|
Number of Signaling Groups | 100 |
Cumulative number of channels* across all the Signaling Groups | 1000 (1 vCPU) 4000 (2 vCPU, 4 vCPU, or 10 vCPU) |
| Call Route Tables | 100 |
Call Route Entries (all Call Route Tables combined) | 1000 |
| Static Routes | 1024 |
Registrar Table entries | 1000 (1 vCPU, 1 GiB) |
Contact Registrant Table entries | 1000 |
| Transformation Tables | 50 |
Transformation Table entries | 1000 |
Media Profiles entries | 20 |
| Callback Number Tables | 80 |
| Callback Numbers (in Callback Numbers List) | 16 |
| SIP Server Table entries | 40 |
| IP/FQDN or DNS-SRV entries (within each SIP Server Table entry) | 99 |
| SIP Message Rule Tables | 100 (32 SIP Message Rules per Table) |
| SIP Profiles | 100 |
*A SIP Channel is a Signaling Group (SG) logical attribute used to represent a potential path for a SIP session (call) leg between the SBC and the remote peer associated with the SG. The SIP channel is not synonymous with a SIP session; the number of SIP channels may equal or exceed the number of supported maximum SIP sessions.
Call Forking
The SBC SWe Lite supports local call forking for up to eight separate destinations. Additionally, the SBC SWe Lite supports up to 20 early dialog responses and 20 calls forked downstream.
SIP Session Licensing
A SIP session for licensing purposes ('SIP session license") is a call (audio/audio+video) under SBC direction. Details are as follows:
- A SIP session is a SIP transaction that establishes a bi-directional audio/video media exchange (RTP media stream) between two ports on the SBC or directly between two SIP endpoints.
- SIP sessions are established by the SBC when the system has recognized the availability of SIP session license "tokens" in the SIP session license token pool.
- SIP session license tokens are added to the pool when a SIP session license has been successfully applied to the SBC. For example, the SBC-1K-LIC25SIP license adds 25 SIP session license tokens to the SBC SIP session license pool.
- To set up the call, the SBC grabs a license token from the purchased SIP session license pool, sets up the call for the bi-directional RTP media stream, and then releases the license token after the call is taken down.
- The media does not have to actually flow through the SBC; the license token is still grabbed to set up the media flow, whether or not the media physically transits the SBC.
- It is possible to consume more than one SIP session license during a single call between two SIP clients. For example, a call that "hairpins" (i.e., one pair of ports supports one RTP media stream through the SBC, and another pair supports a second media stream through the SBC) will consume two SIP session licenses.
SIP transactions that are not directly related to a call setup/tear down are not licensed through the SIP session licenses. Generally, these transactions are free (e.g., SUBSCRIBE, etc.) except when they fall under a chargeable feature. For example, the SBC supports SUBSCRIBE method pass through related transactions (For example, one SIP client to inform another that a message is waiting, etc.) in a way that is limited by available CPU resources, and not by licenses.
For more information on available SIP session licenses available for purchase, and the procedure to apply licenses to a given SBC device, please refer to Working with Licenses.
Burst Behavior
If multiple calls arrive simultaneously, the SBC SWe Lite will service the calls until it reaches a CPU usage threshold (configured raise TCA threshold).
Although TCA can be configured, Ribbon strongly recommends using the default configuration and not changing the default levels. For details on TCA, refer to Working with Historical Data and TCA Thresholds.
- When the threshold is reached, the SBC SWe Lite will reject calls until it reaches a "safe" level (configured clear TCA threshold).
- When the "safe" level is reached, the SBC SWe Lite will begin accepting new calls.
RTP Media Services Supported
For details on supported RTP media services and licensing requirements, please refer to Calculating Virtual Machine Requirements for an SBC SWe Lite.
System Concepts and Terminology
For details on system concepts and terminology, refer to Calculating Virtual Machine Requirements for an SBC SWe Lite.
For additional details on media modes, refer to:
Configuring SBC Edge and SBC SWe Lite for various Media Modes
Overview
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