[asterisk-users] Delay on speak with Asterisk

[Luca Bertoncello]

Am 03.12.2019 um 19:57 schrieb Antony Stone:

Hi Antony,

thank you for your answer.

> I would firstly look at whether your Asterisk box is doing transcoding - 
> converting from oe codec (supported by your phones) and another codec 
> (supported by the provider) because no codec can be found in common between 
> the two.
> 
> Secondly I would put a full packet sniffer (by which I mean collect all the RTP 
> data as well as SIP) on each of your interfaces (internal and external) to see 
> whether the delay really is happening inside your Asterisk server - if you see 
> RTP data on your internal interface, then appearing 1-1.5 seconds later on the 
> external interface, and vice versa, then you know the delay is inside your 
> system.

I'm really not an expert on Asterisk...
Could you please say me HOW can I check the codecs?

I tried to get the information of the channel:

bpi*CLI> sip show channel p65551t1575398506m6025c4749452s2

  * SIP Call
  Curr. trans. direction:  Incoming
  Call-ID:                p65551t1575398506m6025c4749452s2
  Owner channel ID:       SIP/pbxanika-0000021e
  Our Codec Capability:   (alaw)
  Non-Codec Capability (DTMF):   1
  Their Codec Capability:   (ulaw|gsm|alaw|amr)
  Joint Codec Capability:   (alaw)
  Format:                 (alaw)
  T.38 support            No
  Video support           No
  MaxCallBR:              384 kbps
  Theoretical Address:    217.x.x.x:5060
  Received Address:       217.x.x.x:5060
  SIP Transfer mode:      open
  Force rport:            Auto (No)
  Audio IP:               217.y.y.y (local)
  Our Tag:                as45e11359
  Their Tag:
h7g4Esbg_p65551t1575398506m6025c4749452s1_206873930-910452977
  SIP User agent:
  Username:               550293777072-0001
  Peername:               pbxanika
  Original uri:           sip:sgc_c at 217.x.y.z
  Caller-ID:              +4917711111111
  Need Destroy:           No
  Last Message:           Rx: ACK
  Promiscuous Redir:      No
  Route:                  <sip:217.x.y.z;transport=udp;lr>
  DTMF Mode:              rfc2833
  SIP Options:            timer
  Session-Timer:          Inactive
  Transport:              UDP
  Media:                  RTP

Maybe it helps to find the problem?

Thanks
Luca Bertoncello
(lucabert at lucabert.de)