[asterisk-users] switching from SIP to Skype..or not

[Thufir]

On Thu, 12 Mar 2015 10:04:08 -0400, Andres wrote:

> On 3/12/15 9:39 AM, Ron Wheeler wrote:
>> Your characterization may be true but Skype works much better than SIP
>> when it comes to sound quality.
>>
> SIP is not to blame for this.  Its the audio codec being used. Skype has
> spend a great deal of effort with their SILK codec by making it highly
> tolerant of packet loss and jitter.  The same cannot be said for the
> standard codecs Asterisk uses.
>> I have SIP softphone with Asterisk server and Skype on the same
>> workstation.
>> Skype just works better over the same network.
>>

The thing to remember about Skype is that they started out as the small 
guy, and they had some very interesting ideas, IMHO.

I don't actually know it's a sound quality issue, per say.  It's double+ 
NAT, with a wi-fi bridge, plus, sometimes, another wi-fi network.  In 
that situation, skype works from a cell phone!  Granted, there are 
dropped calls, but, eh.

The way things stand, I can't, unfortunately, use Ekiga to connect to the 
**outside** SIP provider because, apparently, there are too many hops:

http://superuser.com/questions/880705/

IAX might be useful in this circumstance :)




-Thufir