I can even start to imagine an incumbent accepting in their facilities a SS7 link coming from a BRI :)
On Dec 9, 2011, at 9:34 PM, Jan Berger wrote: > 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: marc...@m2j.com.br > To: asterisk-ss7@lists.digium.com > Subject: Re: [asterisk-ss7] 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: 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? > > >>> > > >>> -- > > >>> _____________________________________________________________________ > > >>> -- 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 > -- > _____________________________________________________________________ > -- 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
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