I’ve inherited the “telephony engineer” hat (albeit that was not in my job description) and am determined to get this QoS Hell frozen over. The two most frequent customer complaints are dropped calls (that are not specific outbound or inbound calls, nor specific times) and indecipherable conversations that won’t stop breaking up (from beginning to end of call); basically the unendurable, “worse that cell phone” complaints. It doesn’t matter that some complaints come from cable customers with 1.5M/256k lines and some from DSL 5M/512k (and one with fiber); some from Linksys routers with QoS while others with DLink routers with no QoS gasp; some with PolyComm phones and others with SNOM. Basically it’s across the board. The basic customer setup is:
modem <-> router <-> asterisk <-> voip phone
Almost all Asterisk servers also have POTS interfaces which the customers just love. Give them a way to directly tap into an analog line (via an extension) and they try to always bypass the voip lines. Yea, it’s that bad.
My solution is an end-to-end diagnoses of what’s going on with these calls to essentially reverse engineer however they were suppose to work, but I’m not seeing all the tools needed for such testing. There’s these sip testing tools that I just bumped into but from what I’m seeing (via netstat) and these forums, I really need something that will let me spontaneously “make” a phone, aka. softphone, on the far end and on the near end for testing … tracing with wireshark. Any ideas? Otherwise I have to sit around waiting for a session or my other option is – heaven forbid – install etherreal on the server! gasp (These are dedicated asterisk boxes on their end. The other stuff running is … well ntpd and sshd. hehe … BUT, they are merged networks. The data and voice are not separated in any way. And yes, I’ve argued for separate networks or layer2/3 switches … but have been told that throwing money at this is not an option.)
Also, this pseudo-random rtp port numbering seems hell for QoS. I’ve watched asterisk open ports from as low as 11678 to as high as 32345. Some routers don’t support ranges of ports and customers tend to be unconvinced that buying a better, – read “more expensive router” – will help them. Is there a way to tell Asterisk, “hey, you’ve only got 6 voip lines, so you only get 12 ports to play with, and they are 10000 to 10011.” That would be fabulous … especially for the cheap routers playing alongside other programs.