E1 0th channel is framing and alarm. On T1 framing and alarm uses extra 8kbps (T1 raw speed is 64*24+8=1544 kbps). Signaling on E1 usually goes on TS 16, for MFC R2 and ISDN. ISUP signaling can go on any TS.

Unfortunately SIGTRAN is not supported by any Asterisk SS7 we discuss here (if any at all). Before supporting SIGTRAN, it would be useful for many libss7 users to get full ss7 signaling routing (allowing a pair of signaling likes to 1 or 2 STPs to access dozens of switches, with proper failover handling). That's much simpler to implement than SIGTRAN.

Here in Brazil, all interconnects with land line carriers are done exclusively with E1, since we need the E1 for voice, the only cost is taking one TS for signaling.

Apparently cell phone carriers here allow for SS7 over IP interconnects, but then you need full IP interconnects, including supporting their voice codecs (gsm half rate, gsm full rate, enhanced full rate, 3g voice codecs). Voice is then transferred using RTP/IP. Asterisk is very far from being able to support that, even if you disregard the codecs other than gsm full rate which aren't supported with Asterisk.

Usually any soft switch which is able to interconnect using pure IP SS7 is also able to talk SIP and/or H.323.

On 12/10/11 09:06, Michael Mueller wrote:
On Sat, Dec 10, 2011 at 12:11 AM, Paul Timmins<p...@timmins.net>  wrote:
My switch does SS7 over T1 timeslots, but only supports 56k. I'd imagine
it's because DDS modems were used over copper pair for the original links,
and 56k timeslots are how DDS is transported over a T1. (Yes, you can have
64k DDS links, but I've never seen em)
56 kb/s channel speeds in T1 comes from
http://en.wikipedia.org/wiki/Robbed-bit_signaling

in E carriers, the signaling was carried in the 0th channel and that
why only 31 channels are available instead of 32

robbery in both E and T

Also, the number of voice channels you can run over 56k SS7 links is
limitless, the call setup and teardown is what matters, not the number of
standing calls.
voice channels supported between two unique point code pairs is
limited by the size of the CIC field:

// ITU
../common/ckt_def.h:#define MAX_CIC                             4095
// (2^12)-1
// ANSI
../common/ckt_def.h:#define MAX_CIC                             16383
// (2^14)-1

call setup/teardown rate depends on bandwidth of the linkset

each link in the linkset is 64 kb/s or 8kB/s; running at 40% max
utilization 3200 B/s; each call is estimated to be 160 B = (80 B IAM +
4*20 B ACM/ANS/REL/RLC)); 3200/160 = 20 calls per second

But I wanted to point out here that nothing stops you from using ANSI SS7
the same way you describe ITU SS7, there's technically nothing more or less
efficient about it. In fact, the north american setup is actually more
efficient, because you can signal an entire network over two 56k links to
your matched STP pairs, and that's all you need. I'm signaling trunks to
something like 10 tandems, with some trunkgroups as large as 20 DS1s,  in 4
LATAs over just two redundant 56k links to one STP pair. Contrast that to
your individual links per trunkgroup and tell me what's more efficient.
The ANSI network evolved to serve a much larger network than the PTT
networks using ITU networks.  I'd argue that meshing makes more sense
for interconnecting many independent networks, and hub/spoke works
well for a large monopoly needing to defend its position.

A links to the hub/spoke network are expensive for small operators and
have the effect of stifling low end competition.  SIGTRAN access
didn't offer any cost advantages to operators in A link markets.

-Paul



On Dec 9, 2011, at 3:57 PM, Marcelo Pacheco wrote:

Typical North America SS7 signaling links use a dedicated v.35 link. STPs
and switches come with V.35 interfaces for signaling instead of using T1
timeslots.
Today the US basic digital links are 56kbps, I think 64kbps links never
caught up, due to RBS signalling.
In some ways, the North America way of doing things is much less efficient,
but its the way its done.
The true reason traces back to old times, when signaling links ran on
separate analog modems, and voice trunks were still analog, and signaling
links might run at 2400bps or lower speeds ! Those links were terminated to
the switches using v.35 interfaces, and speeds moved up to 56kbps, still
using those v.35 interfaces. The advantage is the same physical interface
can run at higher speeds (56kbps - 1544kbps and in between).

In telco interconnect scenarios, multiples of two 56kbps links are used.
Usually between a pair of STPs on both sides, a small interconnect could
start with just two links, growing to four links. 4x56kbps links are
typically enough for around 4000 voice channels, even considering a 50%
failure.

That contrasts with the extensive utilization of semi permanent digital
calls, using 64kbps timeslots on E1 land.
E1 land makes it so much easier. Just take time slot 16 of an already
existing voice trunks, and switch those time slots to STPs on both sides.
This makes the transport network a lot easier, interconnections only need E1
links between TDM switches, and on each side each TDM switch uses time slots
on existing E1 links to STPs.

The term SS7 on BRI makes no sense. BRI lines are 144kbps (2x64kbps bearer
channels + 16kbps signaling link). Those are never used as SS7 transports.
BRI lines are switch to end user facilities, BRI lines never run between
carrier switches or STPs.

Thanks for listening to my history lesson, useless rant.

Marcelo

On 12/09/11 15:36, Jan Berger wrote:

http://en.wikipedia.org/wiki/Digital_Signal_1

I believe it's BRI lines that uses 56kbps and your right that SS7 on BRI
have some usage in US.

Jan
Date: Wed, 7 Dec 2011 20:12:22 -0200
From:marc...@m2j.com.br
To:asterisk-ss7@lists.digium.com
Subject: Re: [asterisk-ss7] SS7 + T1 on Asterisk?

Typically T1 (american) signaling ss7 links run at 56kbps instead of
64kbps.
If your switch can run 64kbps links over a T1 timeslot, than the only
remaining variable is ITU versus ANSI ISUP. They are incompatible
(different message formats due to different network address sizes and
other details).
We use ITU ISUP all over the place without trouble. If the switch can do
64kbps links and ITU ISUP, then you should be able to use all existing
E1 direct connection samples (without STP), except for the obvious E1=31
timeslots while T1=24 timeslots difference..
ANSI might work. I won't go there because I have zero experience with
ANSI SS7/ISUP (stability wise).
With 2 T1 and a single signaling link it should allow for 47 voice
channels and one signaling link.

Search for libss7 ansi 56kbps for the most difficult scenario. But if
you can do ITU ISUP + 64kbps links, I would suggest that instead.
We hardly see people talking about ANSI ISUP setups on this list, so it
could be far less stable (at least it seems to get less usage).

On 12/07/11 16:25, Matt wrote:
In this case, our supplier is ourselves. We have a DMS100, but the
switch guy is someone other than myself - I am the IP guy.

So basically if I understand you properly, I should be able to do the
SS7+T1 and get proper operation, provided the configuration on both
sides is right.

On Wed, Dec 7, 2011 at 1:06 PM, Marcelo Pacheco<marc...@m2j.com.br>
wrote:
If the DMS100 switch can talk signalling directly with Asterisk,
without an
STP, it should be possible to use a single timeslot for ss7 signalling,
so
with 2 T1 you could have 47 voice calls and one signalling channel.
This is
common with E1 setups. Also with E1 its common for a timeslot to be
statically switched over to an STP (semi permanent call), allowing for
access to the signaling network without a dedicated physically separate
signaling link, but that's not usual in T1 land.

But what you ask is technically possible... However its important to
PROPERLY LEARN SS7 terms to be able to communicate with your supplier.
SS7 is a CARRIER LEVEL PROTOCOL. However people insist on winging it
without
proper training.
Its like trying to become a backbone internet provider without properly
learning inter and intra network routing protocols (like BGP and OSPF).

If you knew the general SS7/ISUP knowledge, you could quickly find the
information you're looking for on Google.

PS: I live in E1 land... I'm just quoting information from the top of
my
head. I have no need for T1+SS7. E1+SS7 is a little simpler with
Asterisk
than T1+SS7 due to 56kbps data links, ANSI ISUP/SS7 and some other
quirks.

Good luck. You'll need it.


On 12/07/11 14:47, Matt wrote:
If I were to get a 2 span T1 card for Asterisk... and connect it to a
Nortel DMS100... can I run call traffic over the T1 and run SS7
signaling FOR the T1 over the other port?

Is there documentation on doing this anywhere?

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