Asterisk SIP

Asterisk Asterisk Support
1

Hello,

I have setup an asterisk system 11.2.1 on a Fedora FC18 box.
The hardware is PC-engines Alix (this one)

I understand I do not have to expect a lot of performance, but it should be okay for a couple (2-3) of SIP phones.

But the lag is 1-5 seconds between my software SIP phones (Vippie on Android 4.2) and my hardware SIP phones (Grandstream GXV3175, 2N Vario IP helios)
Is this normal? if yes, I will stop whining.
Whenever I start a call, the load on the server does not exceed 0.8…

Any hints? Documents I should read?

sip.conf:

[general]
context=external-SIP
allowtransfer=no
;alwaysauthreject=yes
allowguest=no

; Realtime stuff ;
rtcachefriends=yes
rtsavesysname=yes
rtupdate=yes
rtautoclear=yes


; network ;
bindaddr=192.168.102.16
udpbindaddr=192.168.102.16
tcpenable=yes
tcpbindaddr=192.168.102.16
localnet=192.168.101.0/24
localnet=192.168.102.0/24
localnet=192.168.103.0/24
localnet=192.168.104.0/24
externhost=whiner.no-ip.org
externrefresh=300
transport=udp
srvlookup=yes
nat=force_rport,comedia

; media handling ;
directmedia=nonat
sdpsession=Asterisk PBX

; domains ;
allowexternaldomains=no
domain=whiner.lan
domain=sip.whiner.lan
domain=asterisk.whiner.lan
domain=192.168.102.16


; codecs ;
disallow=all
allow=g726
allow=alaw
allow=gsm
allow=ulaw
allow=h263p
allow=h264
allow=h263

; regional settings ;
language=en
tonezone=be

useragent=Asterisk PBX

videosupport=yes
maxcallbitrate=384

SIP users are on MySQL (Realtime)

name, context, defaultuser, type, callerid, host, nat, dtmfmode, mailbox
wle, internal-SIP, wle, friend, Me <8740>, dynamic, force_rport,comedia, rfc2833, 8740@default

Extensions is in Mysql as well:

context, exten, priority, app, appdata
internal, 8740, 1, Macro, dial-sip-vm,SIP/wle
internal-SIP, 8740, 1, GoTo, internal,8740,1

Thank you for your feedback.

2

It’s not reasonable.

3

So I’m asking for too much, eh?
Okay. Back to the drawing board…

4

No. I meant such delays are not reasonable. That system should be able to achieve low latencies.

5

Anything I can do to troubleshoot, except from crunching the debug logs?

6

I would use tcpdump to capture packets and analyze the audio (WireShark has excellent tools for this).

And one more thing - is the first 5 seconds of audio delayed (comes with a delay) or are the first 5 seconds of the call cut?

7

No delay at the beginning, and the call is not cut.

But I believe I may have found the culprit.
I originally built this system with asterisk 1.4, and used the option “canreinvite” to bypass the asterisk for the actual call. Apparently it has been renamed to directmedia since 1.6.2, and I overlooked it.