Switching From SIP To Skype..or Not

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Asterisk Users 10 Comments

I’m testing Asterisk at home, crummy connection. Skype works fine for me, but every SIP client, even without using Asterisk, fails to connect.
That’s ok.

Is swapping out SIP for Skype a big deal?

Heh, well, I guess it’s dead:


If I have a really bad connection, can I “downgrade” SIP somehow? I
don’t really need to use to make voice calls. Or, more specifically, quality, echo, distortion aren’t relevant. Just SIP to SIP “hello”.

When I connect to any SIP provider, ekiga, etc, without using Asterisk, I
get “too many hops” errors. While I have another computer on the LAN I
can connect to, it’s not quite the same.

Any thoughts?



10 thoughts on - Switching From SIP To Skype..or Not

  • Stay away from Skype! It is a toxic, proprietary product. The lack of interoperability by design is the antithesis of what a telecommunication system should be about — and the extent to which they have gone to thwart any attempt at interoperability is truly shocking.

    For connecting two Asterisk installations to each other over the Internet, IAX
    is better than SIP — that’s what it was designed for.

  • Your characterization may be true but Skype works much better than SIP
    when it comes to sound quality.

    I have SIP softphone with Asterisk server and Skype on the same workstation. Skype just works better over the same network.


  • 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.

  • Hey all

    We have been working with SIP for years. It has the potential to be better than Skype. It is really all in the implementation. Not all SIP soft clients are equal nor are the networks and computers they are running on. I will not bash Skype. We have tested it and in most cases choose not to use it. It has it’s place and is good for the user that meets it’s specific target demographic. SIP is a sold communications protocol that can communication with codecs of differ audio and video quality levels, and supports industry standard software and hardware endpoints.

    With SIP you get to choose how good your quality is. With Skype Microsoft does.

    It comes down to what do you want to achieve, how much resource do you want to put in to it, and are you committed to a bit more work for a lot more options and better quality, or do you want a quick and easy solution with differing limits. Both solutions have their place. To me SIP vs Skype is like complaining apples and carrots do you want fruit or veggies you get to choose.

    You can choose to agree or disagree with my statements. I hope they are useful to some.




  • Opus was co-developed by Skype and could be used with Asterisk (if support to it was added).

  • 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:


    IAX might be useful in this circumstance 🙂


  • Sorry for the empty message. Pressed the wrong button.

    I have been wrestling with a pretty generic Asterisk configuration
    (version 11.11.0 ) set up with FreePBX. The trunk SIP is setup to allow ulaw,alaw,gsm, Video is disabled. I was using Eyebeam and am now trying Jitsi. Jitsi has a number of codecs enabled – opus, SILK, G722, speex,PCMU, PCMA, iLBC, GSM, G723 and telephone-event The internet connection from the workstation to my internet supplier
    (workstation to firewall/router to speed test server at ISP) tests at
    13MBs incoming 6Mbs outgoing.

    The problem has always been great sound from the other telephone and choppy sound (dropped sound fragments) from me to the caller with only one call going through Asterisk and the network pretty much dedicated to the my workstation.

    This has survived upgrades of everything (firewall, Asterisk server, workstation)

    This has reduced my Asterisk telephone to an answering machine with Skype as my way of actually talking to people. This fixes the sound issues and is actually cheaper since I pay a low monthly fixed cost for Skype access to all North American telephones. Skype does not seem to have an problem traversing the same network even with two way video active or during multi-party conferences (mix of Skype and telephones in the group).

    I would like to have a reliable 2 way conversation using Asterisk but have not found any suggestions about the source of the problem or how to fix it.


  • well, that was my thinking — hardware.

    If you have just a SIP *client*, ekiga, what-have-you, can it connect out with SIP to SIP fine? because, if so, that would be a powerful litmus test. If that test works, that establishes it’s not the network. (Yes, I know you tested bandwidth already, but I’d at least try SIP to SIP
    client to see if it matches Skype.)

    Once you know that SIP to SIP works, speaking for myself, I’d just do a clean install. If you’ve run out of troubleshooting steps, that’s the one to use.