[Asterisk-Users] Sound issues on SIP-SIP calls

[Bjorn O]

Hello all!

 

For several months now we’ve been experiencing a really strange problem with
sound which best can be explained as choppy/stuttery, and with a touch of
echo on top. Basically, parts of a conversation might be choppy, but often
combined with some echo as well. The sound problem can only be heard by us,
not by the other party. But this is the strange part:

 

The problem only occurs periodically, and only towards one single SIP
provider. All our equipment is SIP based (we use Cisco 79XX equipment with
firmware 7.5 and 8.x), and we have no PSTN lines. Since this has been
lasting for months, I don’t know the exact time the problem occurred at
first. I recall everything as fine when I ran Asterisk 1.0.7, but through
1.0.9, 1.2.2 and 1.2.5 (HEAD) this problem persists. However, I have a hard
time believing this is caused by a bug. Why? Well, as said, we have no
problems with other providers. In addition, we have Cisco phones that
communicate with Asterisk at the office (traffic is switched between two
local subnets) and also over the internet, and all calls made between these
phones are completed without any trouble. 

 

So I tried doing a clean install on a fresh CentOS last week, but with no
luck – the problem is still there. I also checked if there were load or IRQ
issues, which there is not.The box runs APF firewall and has ports
5060-5070, as well as 8000-10000 (for rtp) open tcp/udp. From the phone
information menu I can see that frame size is most likely set to 20 ms,
while the provider uses 20 or 40 ms (not sure).

 

I should add that this provider is among the larger ones in Norway (IP24).
Other users have reported Asterisk to work just fine with it in the
Provider’s local forum, but that is certainly not my experience. So my
question is, is there anything I might have overlooked, or is there a
particular way I can debug this issue?

 

>From sip.conf:

 

[general]

port = 5060

bindaddr = 0.0.0.0

context = default

disallow = all

;allow = ulaw

allow = alaw

;allow = g729

allow=gsm

maxexpirey = 3600

defaultexpirey = 160

externip = xx.xx.xx.xx  // Asterisk’s static IP

;externrefresh = 60

notifymimetype=application/simple-message-summary

dtmfmode= rfc2833

pedantic=no

 

 

register => 11223344:password:11223344 at sip.provider.com:5060/11223344

//phonenumber:password:phonenumber at sip-proxy:5060/extension

 

 

[11223344]

context= full-access

type=friend

secret=apassword

fromuser=11223344

username=11223344

host=sip.provider.com

fromdomain=sip.provider.com

restrictid=yes

canreinvite=no

insecure=very

deny=0.0.0.0/0.0.0.0

permit=123.123.123.0/255.255.255.0 (provider’s subnet)

nat=no

 

Thanks for all help!

 

Regards,

Bjorn


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