Liviu: Created page with " This is a brief introduction on how the PSTN signalling modules are working together in Yate. ===Analog cards or CAS signalling=== Supports FXS and FXO interfaces, signalin..."

2017-10-26T12:56:54Z

Created page with " This is a brief introduction on how the PSTN signalling modules are working together in Yate. ===Analog cards or CAS signalling=== Supports FXS and FXO interfaces, signalin..."

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This is a brief introduction on how the PSTN signalling modules are working together in Yate.

===Analog cards or CAS signalling===

Supports FXS and FXO interfaces, signaling is associated to each channel

* Directly (analog card)
* CAS signaling over TDM (channel banks)
* MGCP gateway with line package

The [http://docs.yate.ro/wiki/Analog analog] module provides the common call control logic.

The lower level [http://docs.yate.ro/wiki/Tdmcard tdmcard] or [http://docs.yate.ro/wiki/Zapcard zapcard] modules provide both voice and access to signaling data.

If [[MGCP_call_agent_module|MGCP]] is used the actual voice is transferred by the [http://docs.yate.ro/wiki/RTP_support_channels yrtpchan] module.

At a minimum you have to configure the '''analog''' channel and one of '''tdmcard''', '''zapcard''' or '''mgcpca''' modules.

The referencing is like this:
Call Controller -> Voice interfaces

===ISDN PRI or BRI===

Supports signaling and voice to be fully associated or not (NFAS)

* PRI or BRI cards with ISDN D-Channel
* D Channel can be backhauled by SIGTRAN IUA over SCTP/IP
* ISDN voice channels on any supported technology

The lower level [http://docs.yate.ro/wiki/Wpcard wpcard] or [http://docs.yate.ro/wiki/Zapcard zapcard] modules provide both TDM voice and access to the signaling D channel.

SIGTRAN IUA can be configured in [[Ysigchan#Configuration|ysigchan]] to [[SigTransport|transport]] the Q.921 signaling of the D channel over SCTP/IP from a Signaling Gateway.

In this case the voice is usually transported over [[MGCP_call_agent_module|MGCP]].

At a minimum you have to configure a Q.931 call controller in '''ysigchan''' and voice circuits in '''wpcard''', '''tdmcard''' or '''mgcpca'''.

If SIGTRAN IUA is configured you must configure the Q.921 level separately in '''ysigchan''' and a matching transport in '''sigtransport''' else the HDLC interface must be defined in the TDM card module.

The referencing is like this:
Call Controller -> Q.921, Voice interfaces
Q.921 -> HDLC Interface (a proper default exists)
Q.921 -> IUA Client (only for IUA)
Q.921 -> Transport (only for IUA)

===SS7===

Supports associated or quasi-associated signaling

* MTP2 on TDM cards
* SIGTRAN M2UA over SCTP/IP to a Signaling Gateway
* SIGTRAN M2PA over SCTP/IP to another SS7 node
* Cisco SLT backhaul
* Signaling Transfer Point functionality (optional)
* ISUP voice channels on any supported technology like TDM and MGCP

If MTP2 is used the [http://docs.yate.ro/wiki/Wpcard wpcard] or [http://docs.yate.ro/wiki/Zapcard zapcard] modules provide access to the TDM signaling link.

SS7 Layer 2 can also be [[SigTransport|transported]] over SIGTRAN of over [[CiscoSM|Cisco SLT]].

Links are configured together in linksets described in [[Ysigchan#Configuration|ysigchan]].

Each linkset defines part of the SS7 network topology but has absolutely nothing to do with voice.

Phone calls are signaled between SS7 points by the ISUP protocol running on top of SS7.

Sections with ''type=ss7-isup'' in the [[Ysigchan#Configuration|ysigchan]] configuration describe each instance defined by the local point code, remote point code and their circuits.

For each such ISUP instance the circuits can be provided over TDM B channels (wpcard, zapcard) or a remote Media Gateway using MGCP.

At a minimum you have to configure an ISUP call controller, a linkset (MTP3) and one or more links in '''ysigchan'''. Voice circuits are defined in '''wpcard''', '''tdmcard''' or '''mgcpca'''.

If SIGTRAN protocols are used you must also configure matching transports in '''sigtransport'''. If Cisco SLT is used for MTP2 backhaul each link must be described in '''ciscosm'''.

The referencing is like this:
Call Controller -> Voice interfaces
Call Controller -> Point Codes (local, remote)

Linkset -> Links
Linkset -> Point Codes (local, adjacent, routes)

Link -> Interface (only if link is MTP2)
Link -> M2UA Client (only if link is M2UA)
Link -> Transport (only if link is M2PA)

M2UA Client -> Transport

Again, there is no direct association between the Call Controller and the Signaling path. The only correlation is established by the Point Codes.

If a Gateway is used it is usually a combination of Signaling Gateway (SIGTRAN, Cisco SLT) and Media Gateway (MGCP).

For both ISDN and ISUP any association between them must be established explicitly by configuration.

Of course it is possible that specific gateways are only Signaling or only Media but normally you need at least one of each.

'''See also'''

* [http://docs.yate.ro/wiki/Analog Analog module]
* [http://docs.yate.ro/wiki/Tdmcard Tdmcard module]
* [http://docs.yate.ro/wiki/Zapcard Zapcard module]
* [[MGCP call agent module]]

Signalling - Revision history

Liviu: Created page with " This is a brief introduction on how the PSTN signalling modules are working together in Yate. ===Analog cards or CAS signalling=== Supports FXS and FXO interfaces, signalin..."