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:
in the case of 120Ohm impedance, there are two twisted pairs, one is Rx while the second is Tx. In 75 Ohm case, you can expect to find two coaxial cables. Data is transferred by means of bi-polar pulses. Line coding is HDB3, that carries data and allows 2.048MHz clock recovery.
Data is split into 256 bits blocks, called frame: 32 timeslots with 8 bits each. 16 frames are grouped into multiframe. Zero timeslot cross the multiframe has a pre defined structure, it is used for synchronization and may also provide several low-speed data channels. Any other timeslot can be observed as 64k data channel. In some cases (like R2 signaling) sixteenths timeslot also has a special structure; in the case of ISDN PRI, it carries signaling.
In the case of circuit-switched calls, timeslot carries G.711 A-law packed data. It means that 13-bits linear PCM samples are compressed into 8 bits, and some mask is xor-ed to avoid transmission zeros or ones during silence.
American T1 differs both in line coding and framing. Voice is packed in accordance with G.711 u-law.
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.
MTP2 (message transfer part) is a link layer. The goal is to deliver messages between two points that are linked directly. Data link can be ether low-speed (LS) 64k or high-speed (HS), 2M.Bit stuffing is used to split bit stream into messages, procedures are defined to establish link and acknowledge messages delivery.
MTP3 provides a mechanism to deliver messages inside a network. Addresses are 14-bits signaling point codes. There are procedures defined for load-sharing, to control routes etc
ISUP (abbreviated from ISdn User Part) is responsible for call control. In most cases it uses MTP3 services only, but there migth be cases when it requires SCCP.
SCCP (signaling connection control part) provide a mechanism to gateway signaling messages between networks. As an example, SIM card numbering (so-called E.212, or IMSI) can be used for message routing. This subsystem provide both connection-oriented and connection-less services are supported.
TCAP (transaction capabilities application part) combines both transaction and component functions.
MAP (mobile application part) is a key subsystem into GSM world, allowing all core network data exchange procedures.
IN and CAMEL (or CAP) are intelligent network application layer protocols.
BSSAP is signaling used between MSC and GSM radio subsystem.
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
MTP2, MTP3 in any case
ISUP when acting as a “SIP Gateway”
SCCP (class 2,3) and BSSAP (“A-Interface Gateway”)
M3UA or M2UA (“SS7 SGW” or “SS7 Signaling and Media Gateway”)
RTP if there is any media (“SIP Gateway” or “SS7 Signaling and Media Gateway”)
MGCP (“SS7 Signaling and Media Gateway”)
SIP (“SIP Gateway”or “A-Interface Gateway”)