SRX Series Integrated Convergence Services Media Gateway Overview
This topic gives an overview of the SRX Series media gateway (SRX Series MGW) and the features that it provides.
Integrated Convergence Services provides a standards-based Session Initiation Protocol (SIP) media gateway (SRX Series MGW) that connects SIP and time-division multiplexing (TDM ) networks so that calls can be made from and routed to local analog telephones, fax machines, legacy PBX (Key) systems, and SIP phones within the branch and across PSTN or SIP trunks.
The SRX Series MGW includes the following features:
- An onboard DSP that accelerates and offloads media processing tasks from the SRX Series MGW’s main CPUs, resulting in increased performance.
- Onboard telephony foreign exchange station (FXS) and foreign exchange office (FXO) POTS interfaces that provide local number preservation for incoming calls and support for emergency calls. If the SRX Series survivable call server (SRX Series SCS) component of Integrated Convergence Services is configured, these interfaces can also be used for call routing when the central SIP peer call server cannot be reached because of network failure conditions or other fault conditions.
- Telephony interface expansion available through Mini-PIM interface card options, includes a 2 Port FXS + 2 Port FX0 Mini-PIM, a 4 Port FXO Mini-PIM, a 4 Port FXS Mini-PIM, and an IP Flex T1/E1 Mini-PIM.
As Figure 1 shows, an SRX Series Services Gateway device running Integrated Convergence Services can provide voice support for analog and SIP phones to and from a SIP peer call server, to and from a SIP trunking service provider acting as a peer proxy server, and to and from the PSTN using various analog and SIP endpoints, including serving as a front end to a PBX device.
Figure 1: SRX Series Services Gateway Running Integrated Convergence Services

The SRX Series MGW and the SRX Series SCS support the following RFC standards:
- SIP protocol transports: UDP and TCP (RFC 3261–Session Initiation Protocol)
- RTP: RFC 3550 (A Transport Protocol for Real-Time Applications)
- Voice codec support: G.711μ, G.711-A, G729 AB for voice encoding and decoding handled by the DSP media processor.
Integrated Convergence Services supports the following features for analog fax machines:
- Codec support to enable analog fax to be sent over G.711 encoding.
- Support to directly connect an analog fax machine to an FXS port, referred to as direct mapping, bypassing auto attendant.
- SIP registration of FXS in which each FXS port may be registered to the primary SIP peer call server via the SRX Series MGW.
Integrated Convergence Services supports the following voice-specific features:
- Voice activity detection (VAD). With VAD, the router sends RTP packets only when activity is detected. Use of VAD maximizes available bandwidth by minimizing the number of packets sent when calls are idle.
- Comfort noise generation transmitted instead of tones when a call is pending or placed on hold and when the parties are not speaking.
- Jitter buffer handling in which a jitter buffer is used to provide local packet caching. This feature allows VoIP packets to be transmitted at a steady rate by reducing jitter.
- Voice continuity to provide for an improved voice quality experience.
- Tone detection to enable calling features based on digit identification or tones generated by other analog devices or the PSTN.
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