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E1/T1 Codec companding (G711ulaw or g711alaw ?)

Hello Guys, 

 

I'm a little confused about what companding methood should an E1 uses. (I know that E1 and T1 uses PCM / G711, but alaw or ulaw?)

As far as I know, default T1 uses G711ulaw and E1 should use G711alaw....

 

I have an E1 configured on my router, but somehow it is using G711ulaw by default, and not alaw.. (I live in Venezuela)

Where do you configure the compading methood you want to use? Does it matter if my E1 is CAS or CCS (ISDN)? (In both cases, I dont know how to choose the compading law but its always using ulaw.. )

I was reading about it and I saw a "compand-type" command under the voice ports, but I also read that "cptone" under the voice port and "cas-custom country" under the controller defines the used law..

 

Here I have two configs that I usually use, none of them says anything about the companding methood but somehow it always uses the ulaw: 

 

--------------------------------------------------

E1 (CCS) //////////////// ----------------->

-------------------------------------------------

 

controller E1 0/1/0
 framing NO-CRC4 
 line-termination 75-ohm

 ds0-group 1 timeslots 1-15 type r2-digital dtmf dnis

 ds0-group 2 timeslots 17-31 type r2-digital r2-compelled ani

 cas-custom 2

  country venezuela

  ani-digits min 4 max 12

 l
voice-port 0/1/0:1

voice-port 0/1/0:2

 

 

 

---------------------------------------------------

E1 (CCS) //////////////// ----------------->

-------------------------------------------------

controller E1 0/0/0

 line-termination 75-ohm
 pri-group timeslots 1-31

interface Serial0/0/0:15

 no ip address
 encapsulation hdlc
 isdn switch-type primary-5ess
 isdn incoming-voice voice
 isdn send-alerting
 isdn sending-complete
 no cdp enable

voice-port 0/0/0:15

---------------------------------------------------------------------------------------

In both cases I'm using pots dial-peer so as far as I know you can't choose codecs on those Dialpeers, only on VoIP.. 

I have no transcoding and I'm using G711ulaw for my VoIP Dialpeers

Thank you for your help! 

2 Accepted Solutions

Accepted Solutions

David,

The compand will be a-law because you are using E1, if it was T1 is would default to u-law..

I think you should set your cptone to ve to ensure the correct cadence,ring & tones.

Usually I have only needed to change to compand type on ISDN-BRI voice ports.

 

Hope this helps.

Regards

Alex

Regards, Alex. Please rate useful posts.

View solution in original post

David,

Dont confuse what codec is being used on the ISDN-E1 with what VOIP is using.

A transcoder is required to bridge VOIP streams with different codecs.

A GATEWAY brides VOICE streams between different formats (technologies)

 

 

PHONE ----IP ------ GATEWAY ----- E1------- PSTN
 |--------G729-------------|        |-----G711-----------|

Remember the gateway's job it to transpose between
Traditional TDM to the E1 packetise into ip (voip)
towards the phones etc.

This is the task carried out by the PVDM(DSP) in your
gateway.

So your VOIP g729 is taken back to it analoge form
then re-encoded into a digital format (using g711) 
to send back to the PSTN.

Hope this helps
Regards
Alex 

Regards, Alex. Please rate useful posts.

View solution in original post

7 Replies 7

acampbell
VIP Alumni
VIP Alumni

David,

 

Under your voice-ports

 

!

voice-port x/x/x

cptone ve

!

 

VE = Venezuela

This should set the correct COMPAND,ring, cadence, tones etc for your country

Regards

Alex

 

Regards, Alex. Please rate useful posts.

Hello Alex...

Thank you for your reply.. 

At this time, everything is working fine.. I have no cptone configured on my voice port.

When I do a show voiceport 0/2/0:15 this is what i get:

ISDN 0/2/0:15 Slot is 0, Subslot is 2, Sub-unit is 0, Port is 15
 Type of VoicePort is ISDN-VOICE
 Operation State is DORMANT
 Administrative State is UP
 No Interface Down Failure
 Description is not set
 Noise Regeneration is enabled
 Non Linear Processing is enabled
 Non Linear Mute is disabled
 Non Linear Threshold is -21 dB
 Music On Hold Threshold is Set to -38 dBm
 In Gain is Set to 0 dB
 Out Attenuation is Set to 0 dB
 Echo Cancellation is enabled
 Echo Cancellation NLP mute is disabled
 Echo Cancellation NLP threshold is -21 dB
 Echo Cancel Coverage is set to 128 ms
 Echo Cancel worst case ERL is set to 6 dB
 Playout-delay Mode is set to adaptive
 Playout-delay Nominal is set to 60 ms
 Playout-delay Maximum is set to 1000 ms
 Playout-delay Minimum mode is set to default, value 40 ms 
 Playout-delay Fax is set to 300 ms
 Connection Mode is normal
 Connection Number is not set
 Initial Time Out is set to 15 s
 Interdigit Time Out is set to 10 s
 Call Disconnect Time Out is set to 60 s
 Ringing Time Out is set to 180 s
 Wait Release Time Out is set to 30 s
 Companding Type is A-law
 Region Tone is set for US
 Station name None, Station number None
 Translation profile (Incoming): 
 Translation profile (Outgoing): 
 Voice class called number pool: 
 lpcor (Incoming): 
 lpcor (Outgoing): 

 

Now, my question is... why is the companding type in A-law if the cptone is US by default? I have no compand-type commands configured at all.. 

As far as I know, US uses U-law

Even if I configure cptone for another region, it wont change unless I change the compand type manually.

Another question would be, when I do a show call active voice br, It says that I'm using G711ulaw in both call legs, the one comming from the PSTN and the one going from my gateway to my CUCM and the phone... Why does that happen if Im using an A-law companding type?

 

 

 

 

David,

The compand will be a-law because you are using E1, if it was T1 is would default to u-law..

I think you should set your cptone to ve to ensure the correct cadence,ring & tones.

Usually I have only needed to change to compand type on ISDN-BRI voice ports.

 

Hope this helps.

Regards

Alex

Regards, Alex. Please rate useful posts.

Alex, 

I think you are right about that, but I'm still a little confused about something... 

I am calling from my cellphone via PSTN to an extension on my CUCM. 

The first call leg should be formed when the call comes in, it is recieved on a pots dialpeer and it should be using G711 as codec, right? If I do a show call active voice brief It should say G711alaw...

The second call leg is formed when my gateway calls to my CUCM and to my extension... in that case, the call goes through an outgoing voip dialpeer which have no codec configured, so it should use G729.

Now, if I have no transconding configured, why do I get this when I do a show call active voice br?

Total call-legs: 2
1700 : 399614 229895054ms.1 +2420 pid:10 Answer 4265150876 active
 dur 00:00:15 tx:782/21896 rx:337/6597
 Tele 0/2/0:15 (399614) [0/2/0.1] tx:15720/6590/0ms g729r8 noise:-67 acom:6  i/0:-50/-58 dBm

1700 : 399615 229895064ms.1 +2400 pid:1 Originate 0904 active
 dur 00:00:15 tx:337/6597 rx:782/15640
 IP 10.0.2.177:19862 SRTP: off rtt:3ms pl:12750/0ms lost:0/0/0 delay:60/60/70ms g729r8 TextRelay: off
 media inactive detected:n media contrl rcvd:n/a timestamp:n/a
 long duration call detected:n long duration call duration:n/a timestamp:n/a

 

 

In both cases, it says that I'm using G729 and thats not possible.. E1 is G711.. 

 

 

 

David,

Dont confuse what codec is being used on the ISDN-E1 with what VOIP is using.

A transcoder is required to bridge VOIP streams with different codecs.

A GATEWAY brides VOICE streams between different formats (technologies)

 

 

PHONE ----IP ------ GATEWAY ----- E1------- PSTN
 |--------G729-------------|        |-----G711-----------|

Remember the gateway's job it to transpose between
Traditional TDM to the E1 packetise into ip (voip)
towards the phones etc.

This is the task carried out by the PVDM(DSP) in your
gateway.

So your VOIP g729 is taken back to it analoge form
then re-encoded into a digital format (using g711) 
to send back to the PSTN.

Hope this helps
Regards
Alex 

Regards, Alex. Please rate useful posts.

Oh.. 

Thank you Alex..  actually it does help

I think I get it know.. 

So it is the gateway's job to transcode from PSTN to VoIP right? (No need to configure transconding for this)

And I should only configure dspfarms to transcode when I want to change the codec from a VoIP call leg to ANOTHER VoIP call leg... am I right?

 

The thing that confused me was that when I told the router to show me the active calls, the telephony call leg said "g729" ... 

 

Total call-legs: 2
1700 : 399614 229895054ms.1 +2420 pid:10 Answer 4265150876 active
 dur 00:00:15 tx:782/21896 rx:337/6597
 Tele 0/2/0:15 (399614) [0/2/0.1] tx:15720/6590/0ms g729r8 noise:-67 acom:6  i/0:-50/-58 dBm

1700 : 399615 229895064ms.1 +2400 pid:1 Originate 0904 active
 dur 00:00:15 tx:337/6597 rx:782/15640
 IP 10.0.2.177:19862 SRTP: off rtt:3ms pl:12750/0ms lost:0/0/0 delay:60/60/70ms g729r8 TextRelay: off
 media inactive detected:n media contrl rcvd:n/a timestamp:n/a
 long duration call detected:n long duration call duration:n/a timestamp:n/a

 

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