Telscale SS7 card

Scope of technology


Isup to SIP GW

BSC for low-cost 2.5G BTS



The scope of technology

TDM Side

..from physical layer

It was not long ago when PCM was the key interface in telecom.

In Europe so-called G.703 are widely spread, while T1 are used in North America.

Strictly, ITU-T G.703 determines physical characteristics only, while G.704 defines the logical structure. So, when mentioning 2 Megabit E1, one means the following:

American T1 differs both in line coding and framing. Voice is packed in accordance with G.711 u-law.

.. to signaling

The next step is signaling. Here it is reasonable to discuss SS7 signaling. Below there is a brief description of European SS7 subsystems related to this project.

SS7 is a set of protocols that meets ISO model.










American ANSI version of SS7 significantly differs. As an example, point codes at MTP layer are 24 bits against 14, SCCP does not have Global Title Translation, and IS-41 is used for mobile applications instead of MAP. But there are practical cases where ANSI and ITU-T versions of SS7 are mixed.



At a first glance, SIGTRAN is the same SS7, reflected to ip networks. It is the set of protocols with approximately the same layered structure as above. Several adaptation layers provide the corresponding SS7 primitives.

Sigtran uses SCTP as a transport layer that allows multi-stream data transfer.

As the first goals, M3UA and M2UA can be mentioned.

M3UA (MTP3 User Adaptation Layer) is defined by RFC 4666. The goal is to provide the same functionality as MTP3 does. Routing key may be used to share SS7 resource between several application servers.

The next to be discussed is M2PA that is an equivalent to MTP2 link, and replaces a media for a signaling link, keeping compatibility with the legacy MTP3 management procedure; When M2PA is used, SGW must have a point code.

M2UA provides an extention of MTP2 to a remote node, where SIGTRAN management procedures are used.

Later on SCCP adaptation layer can be supported (SUA), but it will require convenient web interface to manage routing.


RTP is protocol for transferring media. In our case it makes a sequence of udp packets that carries G.711 packed voice. There must be some negotiation before starting RTP transfer, that must be done by means of other protocols, such as SIP (it uses SDP for such sort of negotiations), MGCP or Megaco.


MGCP seems to be the simplest protocol to manage endpoints (ea establish RTP data transfer).


Now SIP seems to be the main protocol in VoIP world. When speaking about telephony applications, it allows both call control and endpoints management by means of SDP

What should be done on board