[asterisk-users] LAN IAX2 trunk bad audio quality vs. LAN SIP trunk good audio quality

[Zoa]

Hello,

Can you try trunk = no ?
How much jitter do you see on the link ?

Zoa

Gareth Blades wrote:
> There should be no noticeable difference between slin, ulaw and alaw so 
> what you have is fine. The problem must be elsewhere.
>
> Vieri wrote:
>   
>> --- On Thu, 5/13/10, Gareth Blades <list-asterisk at skycomuk.com> wrote:
>>
>>     
>>> Show the details on the active
>>> channels when using both methods and 
>>> check what codecs are being used.
>>>       
>> The audio codecs are different:
>>
>>            Type: SIP
>>           State: Up (6)
>>           Rings: 0
>>   NativeFormats: 0x4 (ulaw)
>>     WriteFormat: 0x40 (slin)
>>      ReadFormat: 0x40 (slin)
>>  WriteTranscode: Yes
>>   ReadTranscode: Yes
>>
>>            Type: IAX2
>>           State: Up (6)
>>           Rings: 0
>>   NativeFormats: 0x8 (alaw)
>>     WriteFormat: 0x8 (alaw)
>>      ReadFormat: 0x8 (alaw)
>>  WriteTranscode: No
>>   ReadTranscode: No
>>
>> By the way, I have this in iax.conf:
>>
>> [interboxIAX2]
>> deny=all
>> allow=ulaw
>> allow=gsm
>> type=friend
>> host=192.168.250.111
>> secret=mysecret
>> auth=plaintext
>> requirecalltoken=no
>> qualify=yes
>> context=mycontext
>> trunk=yes
>> username=interbox
>>
>> Shouldn't the channel details report ulaw instead of alaw?
>>
>> Also, if I change [interboxIAX2] and replace ulaw with alaw, the result is the same (I still experience bad audio quality).
>>
>> Maybe I should try slin but how do I "force it"?
>>
>>     
>>> Vieri wrote:
>>>       
>>>> Hi,
>>>>
>>>> I have an audio quality problem regarding IAX2. I have
>>>>         
>>> 2 Asterisk servers interconnected via 2 LAN trunks at 1Gbps
>>> (no nat, no firewall).
>>>       
>>>> One trunk is SIP and the other IAX2.
>>>> Normally, I use IAX2 but have noticed easily
>>>>         
>>> reproducible audio quality problems (voice in/out is OK but
>>> there's a "third" noise overlapping with a "scratchy sound"
>>> as if it were some kind of interference).
>>>       
>>>> So lately I setup calls to go through the SIP trunk
>>>>         
>>> and audio quality is OK (no "third overlapping noise").
>>>       
>>>> This is happening between Asterisk 1.4.31 and a
>>>>         
>>> 1.2.40.
>>>       
>>>> I'm wondering if there's something I can tweak in IAX2
>>>>         
>>> to eliminate this artifact.
>>>       
>>>> Could the IAX2 jitter buffer between 1.2 and 1.4 be an
>>>>         
>>> issue (I believe it's enabled by default)?
>>>       
>>>> Thanks,
>>>>
>>>> Vieri
>>>>
>>>>
>>>>
>>>>        
>>>>
>>>>         
>>> -- 
>>> _____________________________________________________________________
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>> New to Asterisk? Join us for a live introductory webinar
>>> every Thurs:
>>>            
>>>    http://www.asterisk.org/hello
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>>       
>>       
>>
>>     
>
>
>