The Corelatus Blog
E1/T1 and SDH/SONET telecommunications

Finding the bad link in an SDH network using BIP counters

Posted June 15th 2017

SDH/SONET has relatively powerful mechanisms for detecting transmission errors, far better than E1/T1 lines. We can use those mechanisms to figure out which optical link in a network has errors. This post shows which parts of the network can see errors in various cases—and which parts cannot.

SDH error detection and counters

SDH packs data into virtual containers (VCs). VCs can be nested, for instance an E1 is transported using a VC-12 and up to 63 VC-12s can in turn be packed together into a larger container called a VC-4. Each of those containers has parity bits so that the receiver can see if the data was damaged in transit. These parity bits are called a bit interleaved parity (BIP).

Each add/drop multiplexor (ADM) in an SDH network keeps count of the number of errors in each BIP it checks. Normally, the ADM will only check the BIP in a container it unpacks or re-packs. Each ADM also sends reports of the BIP error counts backwards on each link. Such a report is called a remote error indication (REI).

By looking at BIP counters and REI counters in various parts of the network, we can reason about the likely source of transmission errors.

A concrete example: case 1

On the left side of this network, we have 15 E1s. They are transported along the light blue path over to the right side of the diagram:

Part of an SDH network with bit errors

ADM 1

ADM 1 picks up 15 electrical E1s and packs them into 15 VC-12 containers. The VC-12s are packed into a VC-4 and transmitted over a 155 Mbit/s STM-1 optical link towards the operator's core network.

For this case, we're assuming the fiber between ADM 1 and DXC 1 has a problem which causes bit errors. The problem spot is marked with a red cross on the fiber. DXC 1 is in the best position to detect those bit errors since it is directly connected to the fiber with the problem.

DXC 1

DXC 1 is a high-capacity digital cross-connect which can forward data to the operator's 2.5 Gbit/s STM-16 ring. Some manufacturers abbreviate digital cross-connect system to DXC, others use DCS.

The DXC counts two types of parity errors on the incoming fiber: RS-BIP and MS-BIP (Regenerator Section and Multiplex Section Bit Interleaved Parity). Both count an estimate of the number of bit errors in (almost) the whole STM-1. In this example, 212 errors have been detected by both counters, the diagram shows them in yellow boxes.

DXC 1 also transmits the error counts back to ADM 1, as remote error indication (REI) values. They are shown in the yellow box below each node.

DXC 2

DXC 2 cannot report any transmission errors. The DXC's switching granularity is a whole VC-4, so it never unpacks the VC-4 or VC-12s to check the BIPs. DXC 2 just forwards an exact copy of the incoming VC-4.

ADM 2

ADM 2 unpacks both the VC-4 and all 15 VC-12s, so it can report the HP-BIP (High-order Path, Bit Interleaved Parity) and the LP-BIP (Low-order Path) errors. In this example, 210 errors were detected in the HP-BIP and 13 in the LP-BIP.

A more complicated example: case 2

In this example, two ADMs are connected in series on the left, with each one attaching several E1s to the SDH network:

Part of an SDH network with bit errors

ADM 3 has 13 E1s which it sends rightwards. For this example, the bit errors are on the fiber between ADM 3 and ADM 4, marked with a red cross.

ADM 4 has a further 8 electrical E1s. It unpacks the 13 VC-12s from ADM 3, adds the 8 new VC-12s, makes a new VC-4 and transmits the result to DXC 2.

DXC 2 cannot report any errors because it just copies the incoming VC-4.

ADM 2 unpacks all 21 E1s, so it can see that 13 of the LP-BIP counters have errors, but 8 do not. Armed with a map of the network, this is often enough information to deduce where the problem is. There are no HP-BIP errors, however, because the VC-4 arriving here was made by ADM 4, whereas the transmission errors happened before ADM 4.

Practical Complications

All ADMs and DXCs are capable of counting the number of BIP errors down to the level above their switching granularity. But actually finding the value in the manufacturer's command-line or GUI is not always easy. On Corelatus equipment, the values are displayed on the on-board HTTP server, under SDH. They are also available via the CLI and API.

The names for the BIP error counters vary from manufacturer to manufacturer. Sometimes, the counter is named after the byte in the SDH frame the counters are carried on, i.e. B1, B2, B3 and V5 correspond to RS-BIP, MS-BIP, HP-BIP and LP-BIP, respectively.

When Corelatus equipment is used with SONET, it reports the counters using the standard SONET names, e.g. REI-L for what would be called MS-REI in SDH.

Using an optical splitter, you can always connect a Corelatus SDH Monitor 3.0 to an STM-1 link and see the BIP and REI counters at all levels. That provides more information about errors than just looking at a DXC's counters, partly because the Corelatus system can report BIP errors all the way down to VC-12 and partly because the Corelatus system can decode and record the data on E1/T1 timeslots. In case 2, the DXC does not report the LP-BIP errors because the DXC has no reason to unpack the VC-12s it is carrying, but a Corelatus system connected any of fiber links along the whole transmission path will report the LP-BIP errors and, optionally, decode SS7 or other signalling on individual E1 timeslots.

Permalink | Tags: GTH, telecom-signalling, SDH and SONET

What does an improperly placed monitor point do to a signal?

Posted August 19th 2016

There is a standardised way to monitor E1/T1 links. It is defined in ITU-T G.772 and is formally called a "Protected Monitoring Point". A correctly installed protected monitor point doesn't disturb the live link. This blog entry is about what happens if you install a monitor point incorrectly.

A protected monitor point (in theory)

A G.772 protected monitoring point connects to a live line in such a way that the live line is protected from three common fault conditions after the monitoring point:

  1. An unterminated line, for instance by inadvertently leaving a cable unplugged.
  2. A short circuit.
  3. Unintentional transmission, for instance if the monitoring line is accidentally connected to another transmission line, or if monitoring equipment is accidentally mis-configured to transmit.

Here's one way to build a protected monitor point:

G.772 monitor point schematic

There's an important note on the schematic: "max 1m" cable from the tap-in point to the resistors in the monitor point. The shorter the distance from the tap-in point to the resistors, the better. In many cases, it's possible to mount the resistors just a few millimetres from the tap-in point.

A practical lab monitor point

To demonstrate what happens to an E1 signal, we created a live link in our lab, with connections just like in an operator's network. We've used a 1m long cable, the maximum allowed, from the tap-in point to the resistors. We then looked at the signal using a low-cost handheld oscilloscope.

A monitored live link in the lab

The chassis at the top of the photo is a Corelatus Messenger 2.1 with three modules in it. Messenger is normally used as an active part of an E1 network, e.g. to implement a voicemail system. We're only using the center module, which has two ethernet ports and eight E1/T1 ports. The green cable is the ethernet cable used to control the system.

The yellow line is the live E1 link. It goes from port 'pcm1A' to a break-out box with punch blocks and then on to port 'pcm2A'. We can measure the number of bit errors on the live link. The punch blocks are the same as the ones operators have mounted in racks in their distribution frames.

The red line taps in to the live E1 link at the punch blocks. The length of this red line is critical: if it's too long then signal reflections from the 540 ohm resistors at the end of the red line will disturb the live link. Corelatus specifies a maximum length of 1m for this red line, in the above example we've used two 0.5m cables. In examples further down the page, we'll increase that to see how bad things can get. The red line leads to a box containing the monitoring resistors and then a 5m long cable back to the Messenger 2.1, where we've configured the incoming port to expect an attenuated signal. The length of the cable from the resistors to the monitoring equipment is not critical, it can be up to 200m.

Here's what the signals on this setup look like. On the left, the live signal and on the right the monitored signal, which is attenuated by 20dB. Both signals are clean and have the expected shape.

oscilloscope view of a clean E1 signal oscilloscope view of a clean E1 signal attenuated 20dB

The scope indicates the scale on-screen; the horizontal (time) scale on both is the same, the vertical scales are different. We're measuring through an x10 oscilloscope probe in all cases.

Extending the red line to 10m

A 10m long tap line is out of spec. A real installation should not have such a long tap line. We expect the signal to go from the tap-in point, partially bounce from the resistor box and then interfere with the live signal. The time for a signal to travel down a 10m line, bounce, and travel back up again is approximately 100ns. 100ns is roughly one third of a division on the scope image, so we expect the damage to be that pulses change shape, rather than new pulses appearing between real pulses. The scope images show that the live signal (left) is damaged, whereas the monitored signal (right) is hardly affected:

oscilloscope view of an E1 signal distorted by a reflection oscilloscope view of a clean E1 signal attenuated 20dB

Extending the red line to 35m

Using a 35m long cable to tap in will give a reflection after approximately 350ns. The damage to the live signal is severe, but the monitored signal still looks mostly OK:

oscilloscope view of an E1 signal severely distorted by a reflection oscilloscope view of an E1 signal attenuated 20dB

The disturbance on the live link is so severe that the live link fails completely. Framing is lost; moving the live link from status "OK" to "LFA" and several other layer 1 error counters (e.g. code violation) increase rapidly. Another simple indication that the link has failed is that the LEDs on the live link have changed colour to orange, they're visible at the top of this photo:

lab layout of an E1 tapped with a 35m line

Probing E1 lines with an oscilloscope

The proper way to measure balanced E1 signals in the lab is with a two-channel oscilloscope, with one probe on each of the E1's conductors. The scope can then display the result in differential, i.e. A-B, mode, with no risk of accidentally grounding one side of the E1.

Since the Vellemann oscilloscope only has one channel, there's no A-B mode. So we have no choice but to connect the probe's "ground" to one conductor and the probe's tip to the other. This works fine, and there are no concerns about grounding since the scope is normally battery powered.

The Velleman oscilloscope is cheap and this was the first time we used it, so we checked the measurements using a high quality analog oscilloscope, a Tektronix 2445. The measurements agree well.

We were pleasantly surprised by the Vellemann oscilloscope. It's by far the cheapest Distrelec have, but it works nicely. We left it in the default 'auto' mode and just pressed the 'hold' button a few times for each measurement to get a trace with a couple of pulses in it. It's easily small and cheap enough to take along to a site. (Corelatus has no association with Vellemann or Distrelec, except as a customer. We recommend this scope because it is useful in our experience. We're not paid to recommend it.)

Permalink | Tags: GTH, telecom-signalling

Debugging a PDH frequency offset

Posted March 6th 2016

Once or twice a year, I debug a network problem which turns out to be caused by a bad synchronisation topology. Here's how I debugged the most recent one.

Symptoms

The direct way to see a frequency offset in a PDH network is to measure the frequency at different interfaces in the network, preferably to an accuracy of at least fractional ppm. The frame rate on an E1 line is supposed to be exactly 8 kHz. But... normally, we see less direct symptoms.

One indirect symptom is clearly visible in the layer 1 counters. Below is a screenshot of normal layer 1 counters for an E1. In cases where there's a frequency offset, you'll see the 'slip +' or 'slip -' counters increasing at a rate of about one per minute, or faster, depending on how bad the frequency offset is.

E1 layer 1 counter screenshot

Another indirect symptom is visible in the layer 2 counters of some, but not all, protocols. E.g. in SS7 MTP-2, almost every slip causes a packet to be damaged, so you can see the errored signal unit(ESU) counter increase at a rate of a few per minute. ISDN LAPD, on the other hand tends to hide hide slips, especially at low load.

Analysing a recording

In this case, we could see the SS7 MTP-2 ESU counter increase at an abnormal rate on some timeslots, so I asked for a recording of a couple of minutes of one of the SS7 MTP-2 signalling timeslots which had damaged signalling. Corelatus hardware lets you make bit-exact recordings of live timeslots, either through the HTTP interface or using the 'record' program in the C sample code.

With the recording in hand, I started off by playing it back on a reference GTH 2.1 in our lab, effectively making a copy of the live link in our lab. Turning on MTP-2 decoding, here's what I saw:

PacketTime (milliseconds)Packet dump
10 84 9a 00 6c 4d
2134 85 9a 00 b0 17
32107 85 9a b0 17
...
2017921 86 a2 00 b6 80
2122125 86 a2 00 86
...
2937351 86 a6 00 d6 e7
3041540 a6 00 d6 e7

SS7 MTP-2 links in normal operation carry nonstop packets: when there's nothing useful that needs to be sent, they send a five octet fill-in signal unit (FISU) over and over again. FISUs are always five octets long and they always have 00 as the middle octet.

Looking at the packet dump, packets 1, 2, 21 and 29 are all valid FISUs. Packets 3, 21 and 30 are all invalid---they are too short to be valid MTP-2 packets and they have incorrect CRCs (frame check sequence). It's pretty easy to see that each bad packet is just like its predecessor, except that one octet has been deleted.

(Aside: if you look closely, the defect in packet 21 isn't exactly a deleted octet. That's because MTP-2 uses bit stuffing and isn't octet aligned. We'll ignore that for now.)

A missing octet is a smoking gun for 'slips'

Having one missing octet in a packet is a smoking gun for a 'negative slip': different parts of the operator's PDH network are running at different frequencies, which forces layer 1 to compensate by throwing out a byte every so often. The missing octet rules out other possible causes for packet damage, in particular bit errors.

At this point, the evidence is already strong, but we can add one more thing to make it overwhelming: look at the elapsed time between damaged packets. We expect 'slip' events to be periodic. Between packets 3 and 21, the elapsed time is 20018ms. Between packets 21 and 30, the elapsed time is 19415ms.

From the elapsed time, we can estimate the frequency difference in the mis-synchronised parts of the network. An E1 timeslot is supposed to carry exactly 8000 octets per second, which corresponds to 125 microseconds per octet. A deleted octet every 20s or so corresponds to a frequency error of 6ppm.

In this case, we're seeing negative slips. The reverse is also possible. In a 'positive slip', layer 1 repeats an octet every so often.

Fixing the root cause

At the time of writing, I don't know what the root cause is. But I can offer a guess based on experience: part of the operator's PDH network has lost contact with the operator's primary reference clock, probably because of a configuration error in one or more cross-connects or MUXes in the network.

A primary reference clock is supposed to maintain long-term accuracy to 1 part per 1011 (ITU-T G.811). The frequency offset we can observe in the spacing of the slips is about 6 parts per 106 (6 ppm), i.e. many orders of magnitude worse than intended.

Permalink | Tags: GTH, telecom-signalling

Decoding random data

Posted March 13th 2015

If you were to feed a timeslot random data and try to decode that as though it were MTP-2, how long would it take until you get a valid packet?

(Quick answer: it'll happen about once every week or two on a regular 64kbit/s link.)

What does it take for a packet to be valid MTP-2?

If we send random data, then every so often there's going to be a flag. That signals start- and end-of-packet for MTP-2, so random data will look like it has packets. But to get valid packets, there are several more hoops to jump through:

It's possible to calculate how often all of those requirements are met...but it seems easier and less prone to "oh, I didn't think of that" to just try it and see what happens.

Test setup

I used a Corelatus E1/T1 Messenger 2.1 with a crossover E1 cable so that everything it transmits on one E1 port is received on another. I didn't write any code to do this, I just used programs from the C sample code collection, like this:

    
      $ ./playback_file 172.16.2.7 3A 1 /dev/urandom
      // in another window
      $ ./save_to_pcap 172.16.2.7 4A 1 /tmp/random.pcap
    
  

Results

After about 12 hours, the MTP-2 counters looked like this:

InterfaceTimeslotStatusMSUsESUsFISUsLSSUsLoad %
4A 1 no signal units 3 10964728 0 0 0

We got 11 million bad packets and 3 good ones and no FISUs. Looking at the "good" ones more closely, it's obvious that they're not good either. Here's one of them:

    
      $ tshark -x -r /tmp/random.pcap | tail -2
      0000  9e e2 49 ef a1 95 0f b0 7a 34 b6                  ..I.....z4.

      $ tshark -V -r /tmp/random.pcap
      Frame 3: 11 bytes
      Interface id: 0 (4A:1)
      Encapsulation type: SS7 MTP2 (42)
      Arrival Time: Mar 13, 2015 11:36:56.057000000 CET
      Message Transfer Part Level 2
      .001 1110 = Backward sequence number: 30
      1... .... = Backward indicator bit: 1
      .110 0010 = Forward sequence number: 98
      1... .... = Forward indicator bit: 1
      ..00 1001 = Length Indicator: 9
      01.. .... = Spare: 1
      Message Transfer Part Level 3
      Service information octet
      11.. .... = Network indicator: Reserved for national use (0x03)
      ..10 .... = Spare: 0x02
      .... 1111 = Service indicator: Spare (0x0f)
      Routing label
      .... .... .... .... ..01 0101 1010 0001 = DPC: 5537
      .... 0000 0000 1111 10.. .... .... .... = OPC: 62
      1011 .... .... .... .... .... .... .... = Signalling Link Selector: 11
      Data: 7a34b6
    
  

Two things aren't quite right. First, the spare bits are normally set to zero (but Q.703 doesn't require this). Second, the length indicator is 9, but should be 6 for this packet (Q.703 requires checking this; Corelatus GTH accepts it anyway to make it easier to debug installations which use extended sequence number format).

Permalink | Tags: GTH, telecom-signalling, SDH and SONET

I have 47 E1s? How much hardware do I need to monitor them?

Posted January 14th 2015

Updated 6. February 2018 because of capacity improvements in hardware shipped from this date onwards: we have improved MTP-2 decoding capacity from 96 channels to 240 channels.

A common question when starting a monitoring project is "how much hardware do I need to monitor these E1 (or T1) lines?". The final word is always the specifications:
E1/T1 Monitor 3.0
SDH Monitor 3.0

One way to get started is to look at some real-world examples:

Example 1: 47 E1 lines, each with one 64 kbit/s SS7 MTP-2 signalling link

This is a classic E1 setup. Each E1 carries 30 timeslots of voice and one timeslot of signalling. In this example, we'll assume:

To monitor that, we need enough ports to plug all the E1s into, and we need enough capacity to decode the MTP-2 signalling.

Ports: An E1/T1 Monitor 3.0 has 64 E1 receivers (spec. 2.1.1). The site in this example has 47 E1 lines, but we want both directions of them, so we need 2 x 47 = 94 E1 receivers. So we'll need two E1/T1 Monitor 3.0. We can plug the first 32 E1s into one and the remaining 15 into the other.

MTP-2 decoding capacity: An E1/T1 Monitor 3.0 can monitor 240 simplex ordinary 64 kbit/s MTP-2 channels (spec. 2.2.1). The site has 94 channels. So dimensioning is not affected by the MTP-2 decoding capacity.

Conclusion: The site requires two E1/T1 Monitor 3.0.

Example 2: 12 E1 lines, each with sixteen 64 kbit/s SS7 MTP-2 signalling links

Running many signalling links on the same E1 is common at the core of some networks. It's possible to put as many as 31 SS7 links on the same E1, but 16 is more common. As in the previous example, we'll ignore the remaining timeslots and we'll assume both directions are needed.

Ports: The site has 12 E1s. We want both directions. So we need 24 ports. One E1/T1 Monitor 3.0 has 64 (spec 2.1.1), so we have plenty of ports.

MTP-2 decoding: The site has 12 E1s x 16 signalling links x 2 directions = 384 simplex channels of MTP-2. One E1/T1 Monitor 3.0 can monitor 240. So the MTP-2 decoding capacity is the limiting factor. 384/240 = 2.

Conclusion: The site requires two E1/T1 Monitor 3.0.

Example 3: 8 E1 lines, each with one 1984 kbit/s SS7 MTP-2 high-speed-link

One MTP-2 link on an E1 line can run faster than 64 kbit/s by using more than one timeslot. The formal name for this is "ITU-T Q.703 Annex A", but it's often called "high speed link", "HSL", "HSSL" or "Nx64 MTP-2". We've seen this type of signalling in networks built by NSN.

In theory, all multiples of 64 kbit/s from 128 kbit/s up to 1984 kbit/s are possible, and Corelatus hardware can handle all of them. In practice, 31 x 64 = 1984 kbit/s is the most common. As for the earlier examples, we'll assume both directions are wanted.

Ports: We have plenty.

MTP-2 decoding: The site has 8 E1s x 1 signalling link x 2 directions = 16 channels. The site has 8 E1s x 1 signalling link x 2 directions x 31 timeslots = 496 timeslots of signalling.

One E1/T1 Monitor 3.0 can monitor up to 240 channels and up to 248 timeslots. In this case, the number of timeslots is the limiting factor. 496/248 = 2.

Conclusion: The site requires two E1/T1 Monitor 3.0.

Example 4: 8 E1 lines, each with one 1920 kbit/s SS7-on-ATM link

There is more than one way to run SS7 at high speed. Some networks use ATM-on-E1 to transport SS7. This is also sometimes called "high speed link", "HSL" or "HSSL", leading to confusion because the same descriptions are used for "ITU-T Q.703 Annex A". We've seen this type of signalling in networks built by Ericsson.

The most common way to run ATM on E1 lines is to use all timeslots except for 0 and 16. That gives 30 x 64 = 1920 kbit/s. We'll assume both directions are wanted.

Ports: We have plenty.

ATM decoding: The site has 8 E1s x 1 signalling link x 2 directions = 16 channels. The site has 8 E1s x 1 signalling link x 2 directions x 30 timeslots = 480 timeslots of signalling.

One E1/T1 Monitor 3.0 can monitor up to 16 channels of ATM and up to 496 timeslots (spec. 2.2.4). So one E1/T1 Monitor 3.0 has just enough.

Conclusion: The site requires one E1/T1 Monitor 3.0.

Permalink | Tags: GTH, telecom-signalling

MTP-2 Annex A (High Speed Link) signalling

Posted December 26th 2014

Corelatus hardware can decode MTP-2 Annex A signalling. This post shows how to do it and how to look at the results with Wireshark.

Low speed link and two types of high speed link

E1/T1 lines can carry SS7 signalling in three main ways:

Classic MTP-2: A signalling link uses one timeslot of an E1 or T1. This is described in ITU-T Q.703 and it's sometimes called low speed link, because it runs at just 64 kbit/s (or, on some T1s, 56 kbit/s).

MTP-2 Annex A: A single signalling link uses multiple timeslots on an E1 or T1, allowing the signalling channel to run at up to 1980 kbit/s. This is also described in ITU-T Q.703, in "Annex A". It's sometimes called high speed link (HSL) or high speed signalling link (HSSL).

SS7 over ATM: Just like Annex A, this approach allows signalling to run at up to 1980 kbit/s. But it's done by completely abandoning MTP-2. Instead, it uses ATM AAL5 to carry packets.

Capturing "MTP-2 Annex A" using 'save_to_pcap.c'

Corelatus hardware can handle all of the above ways of carrying SS7, plus all the minor variants, but this post is specifically about the second variant, Annex A. The sample code for controlling Corelatus' products includes a C program for capturing packets to a PCap file, suitable for Wireshark. Here's an example of using it to capture Annex A:

./save_to_pcap 172.16.1.10 1A 1-31 mtp2_annex_a.pcapng

Extended Sequence Number Format

Sometimes, packets captured from Annex A seem to make no sense when viewed in Wireshark. You see thousands of 8-byte packets per second, none of which can be decoded by a higher layer. If you look at the counters on the Corelatus hardware, you'll also see that the link has no FISUs.

This happens when the link uses extended sequence number format (ESNF). It's a variant of MTP-2, and in that variant FISUs have a different format, which prevents the normal FISU filter from removing them. 'save_to_pcap' has a switch to tell the Corelatus hardware that ESNF is being used:

  ./save_to_pcap -f esnf=yes 172.16.1.10 1A 1-31 mtp2_annex_a.pcapng

When viewing with wireshark, you need to tell Wireshark (or tshark) to use extended sequence numbers, either through the GUI or through the command line:

  wireshark -o "mtp2.use_extended_sequence_numbers: TRUE"
  tshark -o "mtp2.use_extended_sequence_numbers: TRUE"
Wireshark decoding an MTP-2 packet with ESNF enabled

Permalink | Tags: GTH, telecom-signalling, wireshark

Daisy-chaining SDH/SONET interfaces

Posted October 27th 2014

SDH/SONET Monitor 3.0 has a daisy-chain feature which lets you retransmit the incoming signal. This is useful in two situations. The first is scaling layer-2 decoding capacity by adding more decoding hardware. The second is monitoring without an optical tap---'intrusive' monitoring.

Ad-hoc daisy-chaining

An earlier post briefly mentioned daisy-chaining. We'll look at the same site: it has three SDH links, S1, S2 and S3. Each link is tapped with an optical splitter. It looks like this:

connecting three STM-1 optical links to four STH sub-modules

Each optical link in the example carries 10 LAPD timeslots per E1, that's fairly typical on Abis links. One STM-1, the one labelled 'S2', is fairly uninteresting because it only carries 300 channels of signalling:

  15 E1s x 2 directions x 10 timeslots = 300 simplex channels

one submodule of an SDH/SONET Monitor 3.0 can decode all of that.

The STM-1 labelled 'S1', however, carries 500 channels of signalling, which is more than the 320 channels one submodule can decode:

  25 E1s x 2 directions x 10 timeslots = 500 simplex channels

The solution is to re-transmit the incoming signal to another submodule. That's what the blue lines in the diagram show. We added a short fiber going from one sub-module to another. That lets us process 320 of the 500 channels on one submodule and 180 on the other.

Symmetric daisy-chaining

I called the daisy-chaining scheme above "ad-hoc", because it fairly arbitrarily copies some directions of some links to another submodule but leaves others alone. There's a second way to do daisy-chaining. Here's an example:

symmetrically daisy-chaining through three submodules

The idea is to feed each direction of a link into a separate submodule, and then copy both directions to a chain of submodules using short fiber cables. In this example, I've shown a chain of three subsubmodules.

Symmetric daisy-chaining lets every sub-module in the chain see all input. You can freely choose which sub-module in a chain processes which signalling channels---e.g. you might want to process the two directions of one signalling link on the same submodule.

The upside of symmetric daisy-chaining is that it's conceptually cleaner: every submodule in a chain sees all of the input.

The downside: symmetric daisy-chaining requires more fibers and in some situations also more hardware.

Intrusive monitoring

Normally, Corelatus hardware connects to optical networks via an optical tap. That way, you can be confident that the monitoring equipment won't disturb the live link.

With daisy-chaining, you also have the option of routing a live signal through the SDH/SONET monitor 3.0, i.e. you don't need an optical splitter. That's particularly useful in situations where briefly interrupting a link is acceptable, e.g. during a site survey or while debugging.

N.B.: daisy-chaining stops working when the power is off! That's why we use optical splitters for permanent installations.

How to set up daisy-chaining

Daisy-chaining always works cross-wise, i.e. a signal coming in on P1 (the leftmost SFP) will come out on P2. Daisy-chaining is off by default. Here's how to enable it:


    <enable name='sdh1'>
      <attribute name='daisy_chain' value='true'/>
    </enable>

Here's what the same command looks like from the CLI:


    GTH CLI started. 'help' lists commands
    gth 172.16.1.34> enable sdh1 daisy_chain true
    ok

Permalink | Tags: GTH, telecom-signalling, SDH and SONET

Getting hex dumps of E1/T1 timeslots

Posted May 7th 2014

This post looks at a few simple tricks for understanding the WWW browser-based hex dump from a typical E1/T1 and then moves on to some more complicated things you can do with timeslot data by piping it to the standard (from BSD, but now on most unix-like systems) hexdump tool.

Browser-based hex dumps

The HTTP server built into GTH (and STH 3.0) lets you click your way to a timeslot-by-timeslot hexdump of an E1/T1. It shows 8ms of data, which gives you a rough idea of what's happening on an E1/T1:

timeslot-by-timeslot hex dump, from L1/PCM1A/hexdump

The HTTP server is on port 8888. To get to the hex dump, click 'L1' (at the top), then the E1/T1 you're interested in, for instance pcm4A, then 'hex dump' (at the bottom).

In the screenshot, timeslot 0 has a repeating two-octet pattern typical of an E1 link using doubleframe.

Timeslots 1 and 2 have the default idle pattern for E1 links: hex 54. That's silence, so those timeslots are most likely unused for the moment.

Timeslot 3 has nonstop 3f 3f 3f 3f. Writing that out in binary, 00111111001111110011111100111111, lets you see the pattern of six ones with a zero on either side. That's a flag. ISDN LAPD and Frame Relay links transmit flags between packets. Timeslot 3 probably contains LAPD signalling. There are eight possible bit rotations of the flag: 7e, e7, fc, cf, 9f, f9, 3f, f3.

Timeslot 4 has a repeating six-octet pattern. That's an MTP-2 FISU. Timeslot 4 is almost certainly running MTP-2.

Command-line hex dumps

This section assumes you're comfortable compiling C programs and using a Unix-like operating system, probably one of the BSDs or Linux. (You can do the same thing using Python, if you prefer, there's sample code for that too.)

Corelatus.com has some C sample code. It's also on github. Using that code, you can record a timeslot to a file:


./record 172.16.2.8 4A 1 /tmp/recording.raw

Let it run for 10 or 20 seconds, then stop it with ^C. You can now look at the data using the standard BSD tool 'hexdump' (on Debian, it's in the 'bsdmainutils' package):


tmp >hexdump -C recording.raw  | head -2
00000000  72 f9 d8 76 e5 df d6 fd  dc 5d ff f5 f0 94 57 da
00000010  eb 6e fa e4 7d 90 e4 e0  91 e2 eb ea ed e4 e3 e4

There's no limit to how large such recordings can be, so if you're debugging something, you can leave the recorder running for hours.

Live hex dumps

The 'record.c' sample code can also write to standard output. You can use that to get a live view of a timeslot via hexdump:


c >./record 172.16.2.7 3A 3 - 2>/dev/null | hexdump -C
00000000  3f 3f 3f 3f 3f 3f 3f 3f  3f 3f 3f 3f 3f 3f 3f 3f
*

The example above demonstrates a neat feature in 'hexdump': 'hexdump' suppresses repeated data. So you can leave it running in a window and it'll only produce output when something changes.

Permalink | Tags: GTH, telecom-signalling, C

save_to_pcap now supports PCap-NG for Wireshark

Posted December 15th 2013

Problem: you want to sniff packets from many SS7 signalling timeslots on many E1/T1s at the same time and analyse them with Wireshark.

Until now, the only way to know which packet came from where was to look at the DPC/OPC. Why? Because the PCap file format, used by Wireshark, tcpdump and many other tools, doesn't have any way to keep track of which interface a packet came from.

Solution: PCap-NG is a completely new file format which lets you keep track of which interface a packet came from. Wireshark understands PCap-NG (and, of course, classic PCap).

save_to_pcap

The C sample code from Corelatus includes a program called 'save_to_pcap' which takes SS7 packets from Corelatus E1/T1 and SDH/SONET hardware and translates them to PCap so that Wireshark can read them.

'save_to_pcap' now saves to PCap-NG by default. Wireshark 10.8 (released in June 2012) reads and writes PCap-NG by default. Here's how to capture packets from 8 signalling channels at the same time:

  $ ./save_to_pcap -n 8 172.16.2.8 1A 1B 2A 2B 16 2 load_generator.pcapng
  monitoring 1A:16
  monitoring 1B:16
  monitoring 2A:16
  monitoring 2B:16
  monitoring 1A:2
  monitoring 1B:2
  monitoring 2A:2
  monitoring 2B:2
  capturing packets, press ^C to abort
  saving to file load_generator.pcapng.1
  saving to file load_generator.pcapng.2

I used '-n 8' to force the capture file to rotate after 8 packets. That gives us clean, closed file to look at. Here's what it looks like in wireshark:

screen capture of the wireshark main application window

I've drawn yellow ellipses around the new parts. To get the "Interface ID" column:

  1. right-click on one of the existing column headings
  2. select "Column Preferences"
  3. select "Field type: Custom" and "Field name: frame.interface_id"
  4. Add
  5. OK

There's actually more information in the PCap-NG file. Wireshark shows it in the Statistics/Summary menu. This shows you the full interface names, i.e. "interface 1" is actually E1/T1 port 1B, timeslot 16 and you can also see exactly which GTH the capture came from.

screen capture of the wireshark statistics dialog

You can also use 'frame.interface_id' in filter expressions.

PCap-NG is a nice format

The PCap-NG format is nicely designed, much better than the original PCap format. Had PCap-NG been around 13 years ago, we probably would have made GTH output traces directly in this format. It's flexible enough to let us include all sorts of information, e.g. we could even add layer 1 status changes on a separate "interface".

Wireshark limitations

PCap-NG is still relatively new in Wireshark, so there are few things that will probably improve with time. The ones I noticed are:

Getting the code

The C sample code is here and also on github.

Permalink | Tags: GTH, telecom-signalling, wireshark

Dimensioning SDH/SONET monitoring

Posted February 27th 2013

Updated 6. February 2018 because of capacity improvements in hardware shipped from this date onwards. This affects the LAPD capacity calculation.

This post is an example of how to figure out how much hardware you need to monitor all the Abis signalling at a GSM site which uses STM-1 to transport E1 lines.

The site

The site we're looking at, a BSC, has STM-1 connections to three BTSs: S1, S2 and S3. They have 25, 15 and 23 E1s on them. Here's how we set it up:

connecting three STM-1 optical links to four STH sub-modules

The yellow lines at the top are the original 155Mbit/s STM-1 links going from the BSC to each BTS. There's one fiber for each direction.

The green lines are the output of the fiber taps. They're an optical copy (typically taking 10% of the light) of the original STM-1. Each link has two taps, one for each direction. For this example, we'll assume that the application needs the signalling from both directions.

The blue lines are fibers going from one SDH/SONET probe's sub-module to another. That lets us shift processing load from one sub-module to another.

Ports and modules on Corelatus' SDH/SONET probe

There are two models of the SDH/SONET probe.

The lower one in the diagram is the low-end model: it has one sub-module. A sub-module has two SFP sockets and two ethernet ports. The SFP sockets accept SFP modules which connect to the optical lines (green).

The upper one in the diagram is the full model: it has three sub-modules.

Layer 2 (LAPD) signalling

For this example, I've assumed that we're interested in 10 timeslots of LAPD signalling on each E1. That's the worst-case in normal GSM installations. In practice, there can be fewer signalling timeslots, either because BTSs are daisy-chained or because the BTS doesn't have the maximum number of radio transceivers.

Each submodule can decode 400 LAPD channels. Link S1 has 25 x 2 x 10 = 500 LAPD channels, so one submodule can't process them all. We handle that by running a fiber to the next sub-module. That way, 25 + 7 E1s are handled on the first submodule and the remaining 18 on the next sub-module.

UMTS

On 3G networks, the signalling is often carried as ATM, either directly on STM-1 (or OC-3) or inside E1 lines carried on STM-1. The principles for dimensioning are the same, but the numbers are different.

Permalink | Tags: GTH, telecom-signalling, SDH and SONET

Retroactive debugging: recording signalling for later analysis

Posted December 6th 2012

This post is about saving signalling data for later analysis.

Problem: a subscriber reports that it wasn't possible to call a particular handset two hours ago. Connecting a signalling analyzer now won't help, so the best you can try to do is to reproduce the problem with test calls.

An alternative approach is to save all signalling data for a few days. Then, when you get a trouble report, examine the signalling affecting the subscriber's handset around the time the problem actually happened.

Disk space

Signalling on a single timeslot of an E1 cannot possibly be more than 8 kByte per second in each direction. There are 86400 seconds in a day and you usually save both directions of the link, so there's never going to be more than 140 MByte of data per day per ordinary link, uncompressed.

Real-world links aren't going to be anywhere near that busy. It's more likely that they're only carrying a few megabytes of signalling per day.

Even on high-speed-signalling-links at 1980 kbit/s, it's never going to be more than 5 GByte per day.

Sample code

The C sample code has an example which saves signalling from as many links as you want to a file in PCAP format.

 ./save_to_pcap -m -n 1000 172.16.1.10 1A 2A 16 captured_packets.pcap

'-m' tells the GTH that the incoming signal is attenuated by 20dB
'-n 1000' means that you want a new '.pcap' file after every 1000 packets; the files automatically get a .1, .2, .3, ... suffix.
'1A 2A' tells the GTH that you want packets captured on the E1/T1 interfaces called 1A and 2A. '16' tells the GTH that you want timeslot 16

The PCAP files can then be opened by wireshark, here's an example of doing that.

Permalink | Tags: GTH, telecom-signalling

Decoding signalling on the Abis interface with Wireshark

Posted February 25th 2012

Wireshark can decode the packets on GSM Abis links (and, probably also UMTS Iub). To do it, you need to first capture the data using a GTH, which is basically the same process as described in this entry about capturing data from the Gb interface, except for one difference: you want to capture LAPD, not Frame Relay:


  save_to_pcap:lapd("172.16.2.7", "3A", 1, "abis.pcap").

Wireshark can open the file 'as is', but it only decodes up to L2. To get wireshark to decode everything, go to Edit/Preferences/Protocols/LAPD and then tick the "Use GSM SAPI values" checkbox. Presto, you get the RSL (Radio Signalling Link) protocol decoded for you. Nice.

Standards

The signalling protocol on GSM Abis links is specified in ETSI TS 100 595. ETSI specs are freely available from ETSI, you just have to register with an email address.

Permalink | Tags: GTH, telecom-signalling, wireshark

Live Wireshark captures on Windows

Posted June 15th 2011

Update 23. January 2014: the -c switch is needed to force save_to_pcap to emit data in classic Pcap format, instead of Pcap-NG. Wireshark 1.10.5 doesn't allow Pcap-NG in pipes.

I've written about using Wireshark to look at signalling captured from an E1/T1 before. As of a few days ago, it's possible to do live captures on Windows, like this:

 save_to_pcap -c 172.16.1.10 1A 2A 16 \\.\pipe\ss7.1
 wireshark -k -i \\.\pipe\ss7.1

When you're capturing live, the SS7 packets appear in Wireshark in real-time.

The new version of 'save_to_pcap' program is part of the C examples, the .zip file contains both source (compiles on Unix and Windows) and .exe files.

It's always been possible to do live captures on Unix, you just pipe stdout:

./save_to_pcap -c gth21 1A 2A 16 - | wireshark -k -i -

Permalink | Tags: GTH, C, Windows, telecom-signalling, wireshark

Decoding UMTS (3G) interfaces with Wireshark

Posted May 24th 2010

Many 3G networks use ATM on their internal interfaces, e.g. on the Iub and Iu-PS interfaces. Those interfaces carry both control information (radio environment information, attach/detach messages, location updates) and also subscriber data, for instance IP traffic.

Wireshark understands how to decode those ATM interfaces. Here's an example of an interface sniffed by a GTH. The interface was carrying IP traffic over ATM on an E1 line.

wireshark screenshot

How to tell the GTH to capture an ATM link

To look at a 3G network like this, you need to:

  1. Connect one of the GTH's E1 interfaces to the E1 (or T1) interface carrying the ATM interface. You typically do that at a cross connect panel, using a G.772 monitor point.
  2. Enable the E1 interface you connected.
  3. Tell the GTH to start decoding ATM AAL5 (and/or AAL2) on that interface
  4. Convert the captured data to the file format which wireshark understands, libpcap.
  5. Open the captured file in wireshark. (It's also possible to pipe the captured data into wireshark live, both on Windows and Unix-like OSs).

Taking those steps one at a time, starting with #2:

Enable the E1 interface


  <set name='pcm3A'><attribute name='monitoring' value='true'/></set>

Tell the GTH to start decoding ATM AAL5

IP traffic on ATM is always carried in AAL5. The timeslot arrangement is usually 1--15 + 17--31. A few sites share the E1 with other protocols, this is called fractional ATM. The GTH can handle either scheme.


  <new>
    <atm_aal5_monitor ip_addr='172.16.2.1' ip_port='1234' vpi='0' vci='5'>
      <pcm_source span='3A' timeslot='1'/>
      <pcm_source span='3A' timeslot='2'/>
      <pcm_source span='3A' timeslot='3'/>
      ..
      <pcm_source span='3A' timeslot='15'/>
      <pcm_source span='3A' timeslot='17'/>
      ..
      <pcm_source span='3A' timeslot='31'/>
    </fr_monitor>
  </new>

In this example, the VPI/VCI is 0/5. If you know the VPI/VCI in advance, great. If you don't, the GTH can sniff traffic at the AAL0 interface and show you which VPI/VCI are active on the link.

Convert the captured data

GTH sends out data in a format described in the API manual. Wireshark wants the data to be in libpcap format. save_to_pcap.erl, in the sample Erlang code for GTH can do the conversion, like this:


  save_to_pcap:from_file("/tmp/captured.raw", "/tmp/captured.pcap").

A lazier approach is to let save_to_pcap.erl configure the GTH and start the capture:


  save_to_pcap:aal5("172.16.2.7", "3A", lists:seq(1,15) ++ lists:seq(17,31),
  {0,5}, "aal5.pcap").

The C version of save_to_pcap can currently only convert MTP-2, not AAL5. If you want it extended, send mail (address at top right).

Start up wireshark

Recent versions of Wireshark, e.g. 1.2.7, can decode such capture files out of the box, without any configuration. Finished.

Permalink | Tags: GTH, telecom-signalling, wireshark

Decoding the Gb Interface with Wireshark

Posted March 31st 2010

The Gb interface is part of the packet radio data network (GPRS) in GSM, it sits between the BSC and the SGSN and carries subscriber data headed to and from the internet.

Wireshark understands how to decode the Gb interface, so you can use wireshark to look through data sniffed from a Gb interface by a Corelatus GTH. Here's what it looks like:

wireshark screenshot

How to tell the GTH to capture Gb

To look at a GPRS network like this, you need to do a few things:

  1. Connect one of the GTH's E1 interfaces to the E1 (or T1) interface carrying the Gb interfaces. You typically do that at a cross connect panel, using a G.772 monitor point.
  2. Enable the E1 interface you connected.
  3. Tell the GTH to start decoding frame relay on that interface
  4. Convert the captured data to the file format which wireshark understands, libpcap.
  5. Open the captured file in wireshark. (On unix-like operating systems, including OSX, you can also pipe into wireshark to get a live view of the capture.)

Taking those steps one at a time, starting with #2:

Enable the E1 interface


  <set name='pcm3A'><attribute name='monitoring' value='true'/></set>

Tell the GTH to start decoding frame relay

The Gb interface uses frame relay on E1. Different sites use different configurations of timeslots. One common setup is to use timeslots 1--15. Another common setup is to 1--15 + 17--31. The GTH can handle any setup.


  <new>
    <fr_monitor ip_addr='172.16.2.1' ip_port='1234'>
      <pcm_source span='3A' timeslot='1'/>
      <pcm_source span='3A' timeslot='2'/>
      <pcm_source span='3A' timeslot='3'/>
      ..
      <pcm_source span='3A' timeslot='15'/>
    </fr_monitor>
  </new>

Convert the captured data

GTH sends out data in a format described in the API manual. Wireshark wants the data to be in libpcap format. save_to_pcap.erl, in the sample Erlang code for GTH can do the conversion, like this:


  save_to_pcap:from_file("/tmp/captured.raw", "/tmp/captured.pcap").

A lazier approach is to let save_to_pcap.erl configure the GTH and start the capture:


  save_to_pcap:frame_relay("172.16.2.7", "3A", lists:seq(1,15), "gprs.pcap").

The C version of save_to_pcap can currently only convert MTP-2, not frame relay. If you want it extended, send mail.

Start up wireshark

By default, wireshark decodes frame relay as 'FRF 3.2/CISCO HDLC'. That's not quite what we want. Go to Edit/Preferences/Protocols/FR and change the encapsulation to 'GPRS Network Service'. Now you get full decoding.

Permalink | Tags: GTH, telecom-signalling, wireshark

Audio power levels on E1/T1 timeslots: the digital milliwatt

Posted October 10th 2009

Sometimes, you want to know when the audio on an E1/T1 timeslot has gotten louder than some limit. In a voice mail application, that's useful for catching mistakes such as a subscriber leaving a message but then not hanging up the phone properly---you don't want to record hours and hours of silence. In an IVR application, you might want to keep an eye on the audio level so that a frustrated (shouting!) subscriber can be forwarded to a human operator.

GTH provides a "level detector" to do that sort of thing. You start a level detector on a timeslot, give it a loudness threshold, and it'll notify you whenever the audio on the timeslot goes over that threshold. Here's an example command which notifies you if the power on timeslot 13 of an E1/T1 is louder than -20dBm0:


  <new><level_detector threshold='-20'>
     <pcm_source span='2A' timeslot='13'/>
  </level_detector></new>

The algorithm is:

  1. Take a 100ms block of audio (800 samples)
  2. Square all the samples
  3. Sum the squares
  4. If the sum exceeds the loudness threshold, send an XML event

There are some details to worry about.

The digital milliwatt

The threshold, e.g. -20 in the example above, has to be relative to something. The standard reference power level in telecommunications is the milliwatt. ITU-T G.711, table 5 and 6 defines the sequences which represent a milliwatt:

A-law: 34 21 21 34 b4 a1 a1 b4
μ-law: 1e 0b 0b 1e 9e 8b 8b 9e

Here's what a few periods of the digital milliwatt look like:

small_linear_milliwatt

In this post, the unit 'dBm0' means power, in dB relative to the digital milliwatt, as defined by the sequences above. If you have no idea what dB means, wikipedia has a decent article.

What's the loudest possible sound on a timeslot?

170 is the highest value possible in A-law encoding. It corresponds to linear 4032. That's about 6dB louder than the digital milliwatt.

85 is the smallest value possible in A-law encoding. It corresponds to linear -1. That's about 66dB softer than the digital milliwatt.

The range -66dBm0...+6dBm0 sets an upper bound on the range of power on a timeslot. Then there are other things which further limit the practical range, so you're unlikely to actually use a +6dBm0 threshold in practice, but it's there if you want it.

Is the G.711 definition the best one?

The sequence given in G.711 is a 1kHz sine wave. The sampling rate on E1/T1 is 8kHz, so the reference sequence can be expressed in just eight values. That's nice, but that also leads to small errors, about 0.13dB, because of quantisation.

ITU-T O.133 discusses that problem in detail and proposes a test signal which specfically is not 1kHz (i.e. not a submultiple of the sampling rate). For most practical purposes, 0.13dB doesn't matter and so the simple and robust thing to do is to use the well-defined and well-known G.711 sequence as a reference.

Here's what a few periods of a 1020Hz signal look like. Notice that the samples, i.e. the red crosses, don't appear in the same spot one period later---that way we don't get the same errors over and over again.

1020Hz signal sampled at 125us intervals
1020Hz signal sampled at 125us intervals

Testing pitfall: the .wav header

GTH players plays raw A-law or μ-law data. If you feed a player a .wav file in 8kHz A-law, or μ-law if your network uses μ-law, there will be a very short bit of noise at the start of the playback because the .wav header gets treated as though it were audio.

When testing level detection, especially at quiet levels, that header noise is enough to trigger a detector. Here's a .wav of a 1000Hz sine wave at about -30dBm0:

00000000  52 49 46 46 42 27 00 00  57 41 56 45 66 6d 74 20  |RIFFB'..WAVEfmt |
00000010  12 00 00 00 06 00 01 00  40 1f 00 00 40 1f 00 00  |........@...@...|
00000020  01 00 08 00 00 00 66 61  63 74 04 00 00 00 10 27  |......fact.....'|
00000030  00 00 64 61 74 61 10 27  00 00 d5 c4 f5 f1 f3 f1  |..data.'........|
00000040  f5 c4 d5 44 75 71 73 71  75 44 d5 c4 f5 f1 f3 f1  |...DuqsquD......|
*
00002740  f5 c4 d5 44 75 71 73 71  75 44                    |...DuqsquD|

The first 58 octets (bytes) are the header. If we turn that header into a periodic signal, it's at about -8dBm0, which is fairly loud. With the default period parameter of 100ms in the level detector, that'll cause a false level of about -11dBm0.

The period parameter

The level_detector has an optional parameter, the period. The period sets the size of the audio block the GTH considers when measuring the power. A short period makes the GTH responsive to sudden changes in power on the timeslot, which would be useful in an application such as figuring out which of the people in a conference call are currently talking. A long period averages out the power over a longer time, which is useful in deciding whether a voicemail recording has finished.

The default is 100ms.

Sample files

This .zip file contains sample recordings with 1kHz sine waves at 0, -10, -20, -30, -40 and -50 dBm0. The .wav versions are useful for listening to or importing into an audio editing program to calibrate the level meter. The .raw versions are just plain A-law samples---you can use them with a GTH player.

Permalink | Tags: GTH, telecom-signalling

Capturing SS7 with wireshark or tshark

Posted August 10th 2009

Update 23. January 2014: save_to_pcap now emits Pcap-NG by default, but current versions (1.10.5)of Wireshark don't allow that through pipes. Use the '-c' switch to force classic Pcap.

I often use wireshark to look at SS7 signalling on E1 links. Up until today, I've always done that by capturing the signalling (from a GTH), then converting the captured data to libpcap format and finally loading the file into wireshark.

Someone showed me a better way today: wireshark can read from a pipe or from standard input. That lets me see and filter the packets in wireshark in real time. Here's how to do it, using the save_to_pcap demo program (included in gth_c_examples):

> ./save_to_pcap -c gth21 1A 2A 16 - | wireshark -k -i -
capturing packets, press ^C to abort
saving capture to stdout

The same thing works for tshark:

 >./save_to_pcap -c gth21 1A 2A 16 - | tshark -V -i -
capturing packets, press ^C to abort
saving capture to stdout
Capturing on -
Frame 1 (15 bytes on wire, 15 bytes captured)
    Arrival Time: Aug 10, 2009 20:38:29.388000000
...
   Message Transfer Part Level 2
    .000 1101 = Backward sequence number: 13
    1... .... = Backward indicator bit: 1
    .011 1000 = Forward sequence number: 56
    1... .... = Forward indicator bit: 1
    ..00 0000 = Length Indicator: 0
    00.. .... = Spare: 0
...

Works on *nix and Windows

Piping standard output to wireshark/tshark works on all the *nixes, i.e. linux, BSD, OSX, Solaris. On Windows, things are a bit different, you have to use 'named pipes' instead, like this:

 save_to_pcap -c 172.16.1.10 1A 2A 16 \\.\pipe\ss7.1
 wireshark -k -i \\.\pipe\ss7.1

On some older (as of August 2009) versions of wireshark, possibly in combination with older libraries, the "-i -" switch doesn't work, at least according to google, even though the tshark version works.

Permalink | Tags: questions-from-customers, telecom-signalling, wireshark

Generating DTMF using a 'player' on GTH

Posted June 9th 2009

The GTH can transmit in-band signalling tones on a timeslot. That's useful for testing and for building active in-band signalling systems.

DTMF

The tones transmitted when the subscriber presses a number key on fixed or mobile handset are called DTMF. Wikipedia has an article about it. To generate DTMF, all we really need to know is that there are 16 possible DTMF signals, that each signal is made up of two sine waves of particular frequencies and that sending the signal for 100ms is a reasonable thing to do.

Here's a .zip file with DTMF tones in it. Each file is raw ALAW data, i.e. it's ready for the GTH to play (transmit) on a timeslot.

The GTH has two ways of playing tones. One way is to stream the audio data in over a TCP socket each time we want to play it. I wrote a post about that earlier. The other way is to store the sample data on the GTH and command its playback whenever it's needed. Since there's a small number of different tones (12, or 16 if you want to use the A/B/C/D tones as well) and the tones are short, storing them on the GTH makes sense. To store the tone:


<new><clip name='dtmf5'></new>
(and now send the 800 byte file)

to play the tone later on:


<new><player>
  <clip id='clip dtmf5'/>
  <pcm_sink span='3A' timeslot='19'/>
</player></new>

Sequences of tones

Sometimes you want to transmit a sequence of DTMF tones, for instance to simulate a subscriber dialling a number. The GTH lets you start a player with a sequence of tones like this:


<new><player>
   <clip id='clip dtmf5'/><clip id='clip dtmf6'/><clip id='clip dtmf8'/>
   <pcm_sink span='3A' timeslot='19'/>
</player></new>

But that isn't a valid sequence of DTMF tones. Why not? Because DTMF expects a gap between tones. The cleanest way to handle that is to define another clip consisting of just silence and putting it between each tone. A good 'silence' value on E1 lines is 0x54. 60ms (480 samples) is a reasonable length.

Other in-band tones

DTMF in-band signalling is used in pretty much all handsets (telephones), mostly for dialling, but also to navigate menus in IVR systems. But before SS7 became popular, in-band signalling in the form of CAS and SS5 was even used to communicate call setup information between exchanges. GTH can also generate those tones, but that can be the subject of another post.

Permalink | Tags: GTH, questions-from-customers, telecom-signalling

Decoding MTP-3 and ISUP

Posted March 25th 2009

Sometimes, you want to look at the signalling on an E1 and use it to figure out when telephone calls start and stop. In SS7 networks, call setup and tear-down is done by the ISUP layer, which fits in to the SS7 stack like this:

Layer 4: ISUP
Layer 3: MTP-3
Layer 2: MTP-2
Layer 1: MTP-1 (typically an 2Mbit/s E1 or a 1.5Mbit/s/T1)

If you have a GTH connected to the E1 you're interested in, either via a DXC or a monitor point, the GTH takes care of layers one and two. That leaves MTP-3 and ISUP to you.

The easiest way to decode MTP-3 and ISUP is to let wireshark do it for you. There's a note about how to do that on Corelatus' official site. But this blog entry is about how to decode MTP-3 and ISUP yourself.

A signal unit (packet)

In SS7, packets are usually called "signal units". Here's what an SS7 signal unit looks like 'on the wire', octet by octet, with MTP-2 and MTP-1 already decoded:

  8d c8 1f 85 02 40 00 00  35 00 01 00 21 00 0a 02
  02 08 06 01 10 12 52 55  21 0a 06 07 01 11 13 53
  55 00 6e 00

MTP-3

The start of the packet is the MTP-2 (ITU-T Q.703) and MTP-3 (ITU-T Q.704) headers. These headers are easy to decode because they are always fixed-length:

Octet(s) Value Purpose
00--01 8d c8 MTP-2 sequence numbers, safe to ignore
02 1f MTP-2 length indicator. Anything less than 3 is reserved for MTP-2 itself and should be discarded.
03 85 MTP-2 SIO. The SIO tells us which 'service' the signal unit is intended for. Q.704 sections14.2.1 and 14.2.2 tell us that anything ending in hex 5 is for ISUP.
04--07 02 40 00 00 MTP-3 Routing label. The routing label is just a "from" and "to" address in the SS7 network. For most applications we can ignore it. Q.704 figure 3 shows what's in the routing label.

Upshot: to see calls start and stop, all we have to do for MTP-3 is:

  1. Look at the length indicator (offset 2) and discard any signal unit where it's less than 3.
  2. Look at the SIO (offset 3). Discard if (SIO & 0x0f != 5)

ISUP

The rest of the signal unit is ISUP. Annex C in ITU-T Q.767 tells us how to decode ISUP. ISUP is fiddly because there are several types of ISUP packets, because several of those types have optional fields and because some of those fields are variable length. Here are the octets we have left after removing MTP-2 and MTP-3:

  35 00 01 00 21 00 0a 02
  02 08 06 01 10 12 52 55
  21 0a 06 07 01 11 13 53
  55 00 6e 00

The first two octets are the CIC. The third octet is the Message type.

The CIC (Q.767 C.1.2) tells us which circuit this call uses. All the signalling for one call has the same CIC. In ITU networks, it's a 12-bit value packed into the field in little-endian byte order. In this case CIC=0x0035. We're sniffing an E1 line, so C.1.2.a tells us that the lower five bits correspond to the timeslot (timeslot 5) and the rest identifies the E1 itself.

The Message Type (Q.767 Table C-3) field tells us what sort of ISUP message this signal unit is. 0x01 is an IAM. 0x10 is RLC. For a minimal "show me what calls are going through the system" hack, we only need to look at the IAM (comes at the start of the call, contains the A and B numbers) and the RLC (sent when the call is finished) messages.

Now we know that the CIC=0x35, that the message is an IAM and we still have about a dozen octets to decode. Q.767 table C-16 tells us how to decode an IAM. There are some uninteresting fixed-length fields followed by the B number and then the A number. Look at the code (or Q.767, section C.3.7) if you're interested in the details. All we really care about is that these octets

  06 01 10 12 52 55 21

represent the B number: 21255512. You can see the number in the raw data if you skip the first three octets and swap every second digit.

Turning those ISUP steps into an algorithm to decode one signal unit:

  1. Save the CIC
  2. Is the message an IAM? Decode it as an IAM, which is fiddly.
  3. Is the message an RLC? Just print the CIC.

That's all you need to do to make a simple system which prints the start and end of each call. To do something useful, you need to maintain a table of in-progress calls and match up the IAM and RLC messages with the same CIC. You also need to handle things like systems restarting.

Further reading

The ITU now have most of their standards freely available at www.itu.int. So one way to learn more about MTP-3 and ISUP is to read the standards, e.g. all the Q-series standards about signalling are here.

Erlang code

Everything discussed above is implemented in the ss7_sniffer.erl example. It makes good use of Erlang's binary syntax, e.g. here's the MTP-3 decoder:


    mtp3(<<_Sub:4, Service_indicator:4>>, <<DPC:14, OPC:14, SLS:4,
						    Rest/binary>>) ->
	case Service_indicator of
	0 -> % Management
        ignore;
	1 -> % Test/maintenance
        ignore;
	3 -> % SCCP
        ignore;
	5 ->
        isup(DPC, OPC, SLS, Rest);
	9 -> % B-ISUP; similar to ISUP, but not compatible.
        ignore;
	X ->
        io:fwrite("ignoring SU with unexpected service indicator=~p\n", [X])
	end.

It looks a lot like one of the examples in the original paper about the binary syntax.

Python Code

The same thing done in Python is fairly straightforward once you discover the Python 'struct' library, which is basically the same thing as PERL's pack/unpack. The code is in sniff_isup.py, inside the GTH python examples zip.

Comments

Bartosz, 28. January 2010

Hey. Just stumbled on this page while I was googling after Mtp3 procedures. I'm wondering if you have any materials that you could share regarding link setup - I'm working on open source stacks for ss7. Im able to setup mtp2,3 and layer 4(signle link), now I'm working on getting it work with more than one and frankly Qs for mtp3 are not very helpful.

(website)

Matt, 28. January 2010

@Bartosz: I only really work with MTP2. My customers use MTP3, so I know quite a bit about it, but you probably know more. For MTP2, the standard is difficult to read, but comprehensive. MTP3 seems harder. I have a copy of "Signalling in Telecommunications Networks" by van Bosse et al, it's useful for getting started, but I always end up trudging through the standards.

Interesting to see someone working on an open SS7 stack. Up until now, I've only been aware of openss7.org (written in C, some parts seem complete, the project seems to be a one-man effort).

Or: I don't think I can help much, but what you're doing looks interesting.

Matt

Emza, September 15, 2009

I found it very important but still not enough for me. I was looking for ISUP MTP-3 and ISUP decoding in C or in C#(not only start and end singnal unit but general which includes all message types). I would like to thank you but can help in my case please.

Thanks again
Embza

Matt, September 15, 2009

@Emza, decoding the rest of the message types is 'just' a matter of working your way through ITU-T Q.767 (ITU standards are now available for free at https://www.itu.int) and writing code to handle each and every section in the standard. Same basic idea as the messages I did. Doing it all is a few days of drudge work, which is why I haven't done it here.

About doing it in C or C#. I know nothing about C#. I work with C most days, though. Doing this sort of protocol decoding in C is fairly straightforward, but it's inevitably going to be more tedious than doing it in something like python, perl or erlang. In some applications the performance gain might be worth the extra effort.

Meskerem David, September 15, 2009

It is a great job. Keep working...

Can u please me one brief algorithm of decoding ISUP. Especially I am not clear how to decode the variable length parameters and the optional ones.

Thanks
Meskerem

Matt, September 15, 2009

@Meskerem, In an IAM, the A-number and B-number parameters are variable length. The algorithm for finding the B-number start is trivial, it's just a pointer offset. Here's what it looks like in the python example code:

    

      def isup_iam(_, CIC, sif):
      # First 5 octets can be ignored
      bnum_pointer = ord(sif[5])
      bnum = sif[5 + bnum_pointer:]
      print "IAM called party: %s CIC=%d" \
      % (isup_number(bnum), CIC)

      # And here's how the example code figures out the length:

      # Decode an ISUP number, as per C 3.7
      def isup_number(num):
      length = ord(num[0]) - 2
    
  

The A-number is trickier because it's an optional parameter. The python sample code steps through the optional parameters looking for the A-number.

(the site also has exactly the same decoding routines in Perl and Erlang)

(Edit 2017-02-20: Thanks to Moy for pointing out that the code to find the A-number was a nasty hack which assumed that the A-number was the first optional parameter present.)

Permalink | Tags: erlang, GTH, telecom-signalling, python

The timestamp field in signalling headers

Posted March 23rd 2009

When the Corelatus GTH is used to monitor (sniff) signalling, it sends each sniffed packet to your server over a TCP socket, along with a header. For instance, for SS7 MTP-2 the header looks like this:

octet 0x00:  Length (16 bits)
octet 0x02:  Tag (16 bits)
octet 0x04:  Flags (16 bits)
octet 0x06: Timestamp (48 bits)

Every field is big-endian, i.e. the most significant byte comes first. Here's an actual header from a GTH, octet by octet:

00 1c 00 00 00 00 01 20  34 ee fa 61 99 99 99 99 ...

The timestamp is thus 0x012034eefa61, or decimal 1237838658145. For most applications, you just want to know which packet came first, so the interpretation of that number doesn't matter much, though it's useful to know that it's the number of milliseconds since the unix epoch. (wikipedia has a decent article about unix time)

Sometimes, though, you want to represent that as a human-readable time. Unix (and, most likely, Win32) provides functions to do that in the C library, so, after throwing away the last three digits (the milliseconds), this C program does it:


#include <time.h>
#include <stdio.h>

int main() {
  const time_t time_stamp = 1237838658;
  printf("%d corresponds to %s\n", time_stamp, ctime(&time_stamp));

  return 0;
}

The output agrees with what the clock on my wall says:

1237838658 corresponds to Mon Mar 23 21:04:18 2009

Python

Since I've been messing around with python, the same thing in python:


>>> import time
>>> time.ctime(1237838658)
'Mon Mar 23 21:04:18 2009'

Erlang

Erlang doesn't have an interface to the 'ctime' call, but you can use the gregorian calendar functions:


1> Epoch = calendar:datetime_to_gregorian_seconds({{1970, 1, 1}, {0,0,0}}).
62167219200
2> calendar:gregorian_seconds_to_datetime(1237838658 + Epoch).
{{2009,3,23},{20,4,18}}

Why use milliseconds?

Why is the GTH timestamp in milliseconds instead of either seconds or a 'timeval'-like seconds + microseconds?

We chose millisecond resolution for several reasons. Firstly, the shortest possible useful packet in SS7 takes a bit more than a millisecond to transmit at 64kbit/s. Secondly, the practical limit of NTP time synchronisation over the internet is about one millisecond at a typical site.

Permalink | Tags: GTH, questions-from-customers, telecom-signalling