It makes little sense if you think about ISUP and voice transport and the cost of E1/T1 hardware today. But, many SCCP/TCAP applications can manage well with 2x64kbs/1x16kbs links. These days you would just use SIGTRAN or grab a E1/T1, but E1/T1 hardware used to be very expensive so many cost-saving schemes have been used over the years. Jan Date: Fri, 9 Dec 2011 18:57:53 -0200 From: marcelo@xxxxxxxxxx To: asterisk-ss7 at lists.digium.com Subject: Re: SS7 + T1 on Asterisk? 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: marcelo at m2j.com.br > To: asterisk-ss7 at 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<marcelo at 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? > >>> > >>> -- > >>> _____________________________________________________________________ > >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > >>> > >>> asterisk-ss7 mailing list > >>> To UNSUBSCRIBE or update options visit: > >>> http://lists.digium.com/mailman/listinfo/asterisk-ss7 > >>> > >> > >> -- > >> _____________________________________________________________________ > >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > >> > >> asterisk-ss7 mailing list > >> To UNSUBSCRIBE or update options visit: > >> http://lists.digium.com/mailman/listinfo/asterisk-ss7 > > -- > > _____________________________________________________________________ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > asterisk-ss7 mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-ss7 > > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-ss7 mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-ss7 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-ss7 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-ss7 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-ss7 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-ss7 -------------- next part -------------- An HTML attachment was scrubbed... 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