[Asterisk-ss7] Let us design an open source
SS7 interfaceforAsterisk
CARDOSO Jorge Miguel
jmc at ip.novis.pt
Mon May 16 09:08:10 CDT 2005
I will try to explain it better.
For ss7 on PSTN side we need one or dedicated timeslots to handle
signalling, and we can have several E1's/ T1's on Gws. For establishing
calls on VoiP side we can use MGCP between ASterisk and Gateway(s).
For making it happen on PSTN side using ss7 we may use TCP between
Gateway and Softswitch. This avoids having dedicated signalling
hardware, because the feature it's already there, on the Gateways.
Softswitch <---------- MGCP -------> Cisco Gw ---E1 (PCM Timeslot-->
<---------- TCP -------> ---E1 (Sig.TimeSlot
The signalling is directly mapped between TCP <---> E1/ T1 Timeslot.
TCP for SoftSwitch side and E1/ T1 for the PSTN side.
In this kind of scenario, we would just need to deploy the ss7 protocol
stack and interface it with TCP. This would also require coordination
with MGCP (to establish, change and delete voice paths).
This kind of implementation is beeing followed by main vendors.
Best Regards,
Jorge Cardoso
Matthew Crocker wrote:
>
> The D channel in a PRI is a subset of ISUP/SS7. It is cannot handle a
> full SS7 stack or features.
>
> In order for Asterisk to become a true 'soft switch' it needs to be
> able to handle ISUP and TCAP (LNP, 800 dips, ANI-Triggers). The ISUP/
> TCAP can get into Asterisk via 2 methods.
>
> 1). Traditional packet based SS7 where a 56kbps data channel carrying
> SS7 is plugged directly into a port on the Asterisk machine.
>
> For more information take a look at the TDM400P-SS7 Drivers found
> on openss7.org They work with the Digium boards. Asterisk would
> terminate the MTP1, 2 & 3 and gain access to the ISUP/TCAP messages
> carried over the SS7 link
>
> 2). SS7 over IP (SIGTRAN). The ISUP/TCAP messages would arrive at
> the Asterisk box over an IP connection using SCTP. The ISUP/TCAP could
> originate at an SS7 to IP gateway or come direct from an SS7 over IP
> provider. Verisign is already offering SS7oIP gateway service. The
> RBOCs are starting to provide it as well. This is normally not done
> over the Internet but with a private T1 based IP network.
>
> For more information on SIGTRAN and SCTP take a look at http://
> www.sigtran.org and RFC2960 (http://www.ietf.org/rfc/rfc2960.txt)
>
> IMHO, I think option 2 is the best, option 1 can be added later if
> needed. Option 2 gives flexibility in gateways and/or gateway
> providers. It also removes a lot of the headaches from dealing with
> MTP1 & MTP2. Linux already supports SCTP in the kernel (linux 2.4
> module) so the only thing that needs to be done is handling ISUP & TCAP
> (chan_isup, chan_tcap) ????
>
>
> When dealing with SS7 you get a 'point code', (like an IP address).
> You need to plug into the network somewhere, normally it is into an STP
> (think SS7 Router). Once you establish a connection to the network you
> need to build ISUP routes over that network to the various devices on
> the network. Devices include tandems and End Office switches, LNP
> databases, etc... When you need to make a call you first send TCAP
> messages to find the correct LNP information, then you send ISUP
> messages to the proper switch to create the call. When you get a 1000
> block of numbers, you publish them in the LERG with the point code for
> the switch handling the numbers.
>
> With SS7oIP your point code is routed over the SS7 network to your
> SS7oIP gateway. The gateway takes the ISUP/TCAP messages, wraps them
> in SCTP packets and sends them to your switch via IP. One SCTP packet
> can contain more than one message (SCTP data chunk).
>
> Verizon requires that I connect directly to the STP (SS7 routers) in
> each LATA that I want to send/receive calls in. Verisign already has
> connections to all of the Verizon STPs so one connection to Verisign
> (over IP) gets me all of the Verizon STPs. I can then build ISUP
> routes over that to the switches I need. Initially I would only need
> ISUP routes to the access tandem, PSAP tandem and LNP database. I
> would also need IMTs between my media gateway and the access & PSAP
> tandems in each LATA. Once I had enough outbound calls during busy
> hour to fill a DS-1 to a single EO, I'm required to build IMTs directly
> to the EO. I would then need an ISUP route to that EO as well. All of
> the point-code, routing information is contained in the ISUP/TCAP
> messages.
>
> The asterisk 'call flow' the way I see it, please correct me if I'm
> wrong:
>
> PSTN -> Asterisk -> SIP phone call using SS7.
>
> 1). ISUP message comes into Asterisk containing the called number
> information (chan_isup)
> 2). Asterisk determines the correct SIP extension (extensions.conf) and
> sends a SIP message to the phone (chan_sip)
> 3). The phone rings, a customer answers. The phone sends a SIP
> message back to Asterisk notifying it of the answer (chan_sip)
> 4). Asterisk determines the correct IMT/Media gateway needed to handle
> the call (from tandem? from end office?, available trunks...)
> 5). Asterisk sends a message to the media gateway to reserve a certain
> channel/port and assign it to SIP, phone IP ... (chan_mgcp)
> 6). Asterisk sends an ISUP message to the originating switch with
> information on which trunk to send the call down (chan_sip)
> 7). Originating PSTN switch reserves the trunk and connects the caller
> to it
> 8). The Media gateway turns the trunk data into g.729 and sends it to
> the SIP phone.
> 9) <Conversation>
> 10). SIP phone hangs up, sends a message to Asterisk (chan_sip)
> 11). Asterisk sends a message to the media gateway to clean up the call
> (chan_mgcp)
> 12). Asterisk sends a message to the PSTN switch to clean up the call
> (chan_isup)
>
> During the entire call process Asterisk needs to maintain the state of
> the call, and the media gateway IMT trunks
> Asterisk would need to maintain a 'map' of the network containing all
> of the Media Gateways, their trunks and which switches they are
> connected to. MG #1, Trunk #1 connected to Tandem (point code) as
> Trunk #1 in Trunk Group #1 .... Asterisk would also need to maintain
> the relevant parts of the LERG for outbound call routing (NPA-NXX goes
> to EO (point code) which can be reached via the access tandem and trunk
> group #1)
>
> A lot of the work is already out there, as open source. It just needs
> to be put together.
>
> On May 16, 2005, at 8:42 AM, CARDOSO Jorge Miguel wrote:
>
>> Some Cisco Gw's support a feature called "MGCP PRI BackHaul", this
>> means the bearer channels can be controlled using MGCP and PSTN
>> signalling (ss7) can be exchanged between Softswitch and PSTN using TCP.
>>
>>
>>
>> Regards,
>> Jorge Cardoso
>>
>> Race Vanderdecken wrote:
>>
>>> Okay, there seems to be agreement as to how to do this. Doing ISUP and
>>> TCAP is what I have heard from others I have proposed the same
>>> questions
>>> to in the past.
>>> Running the SS7 to PSTN interface on a "cisco" type box also has been
>>> recommended by others. By "cisco" I mean any certified box that is
>>> available to most of the VoIP guys. Having to purchase an X thousand
>>> dollar box is not going to work. Have any of you looked at ss7box.com?
>>> So, we need to create a SS7 socket interface to asterisk.
>>> I have built several socket interfaces in the past that have worked in
>>> commercial products. All were based on Stevens' Unix Programming and
>>> Networking books.
>>> Where do I find the socket interface on our "cisco" ss7 box?
>>>
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>>
>
> --
> Matthew S. Crocker
> Vice President
> Crocker Communications, Inc.
> Internet Division
> PO BOX 710
> Greenfield, MA 01302-0710
> http://www.crocker.com
>
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