[Asterisk-Users] iconnect quality?

[Jim Archer]

I pulled the CVS tree at 11:30PM (approximately) Eastern on March 11.  This 
problem does seem to be fixed.  But there is a new issue.  It now seems 
that adding the 7777 prefix that solved the problems before now causes the 
"488 Not Acceptable Media" error response:

856.137213 192.203.175.9 -> 213.137.73.178 SIP/SDP Request: INVITE 
sip:77771401949nnnn at 213.137.73.178, with session description
856.283972 213.137.73.176 -> 192.203.175.9 SIP Status: 100 Trying
856.284294 213.137.73.176 -> 192.203.175.9 SIP Status: 407 Proxy 
Authentication Required
856.284651 192.203.175.9 -> 213.137.73.178 SIP Request: ACK 
sip:77771401949nnnn at 213.137.73.178
856.285463 192.203.175.9 -> 213.137.73.178 IP Fragmented IP protocol 
(proto=UDP 0x11, off=552)
856.285493 192.203.175.9 -> 213.137.73.178 SIP Request: INVITE 
sip:77771401949nnnn at 213.137.73.178
856.423732 213.137.73.176 -> 192.203.175.9 SIP Status: 100 Trying
856.594346 213.137.73.176 -> 192.203.175.9 SIP Status: 488 Not Acceptable 
Media
856.598273 192.203.175.9 -> 213.137.73.178 SIP Request: ACK 
sip:77771401949nnnn at 213.137.73.178
857.233615 213.137.73.176 -> 192.203.175.9 SIP Status: 488 Not Acceptable 
Media
858.123355 213.137.73.176 -> 192.203.175.9 SIP Status: 488 Not Acceptable 
Media
861.142317 213.137.73.176 -> 192.203.175.9 SIP Status: 488 Not Acceptable 
Media
867.170217 213.137.73.176 -> 192.203.175.9 SIP Status: 488 Not Acceptable 
Media

Removing the 7777 prefix seems to make an improvement.  The called phone 
will ring.  If I answer it, though, Asterisk does not seem to know and 
keeps ringing my phone.  If I call my cell phone and hang up Asterisk seems 
to notice, but if I call a POTS line, pick it up then hang up Asterisj 
seems not to notice:

 56.265536 192.203.175.9 -> 213.137.73.178 SIP/SDP Request: INVITE 
sip:1401949nnnn at 213.137.73.178, with session description
 56.390918 213.137.73.176 -> 192.203.175.9 SIP Status: 100 Trying
 56.391242 213.137.73.176 -> 192.203.175.9 SIP Status: 407 Proxy 
Authentication Required
 56.391571 192.203.175.9 -> 213.137.73.178 SIP Request: ACK 
sip:1401949nnnn at 213.137.73.178
 56.392423 192.203.175.9 -> 213.137.73.178 IP Fragmented IP protocol 
(proto=UDP 0x11, off=552)
 56.392453 192.203.175.9 -> 213.137.73.178 SIP Request: INVITE 
sip:1401949nnnn at 213.137.73.178
 56.521701 213.137.73.176 -> 192.203.175.9 SIP Status: 100 Trying
 59.769661 213.137.73.176 -> 192.203.175.9 SIP Status: 180 Ringing
 64.637854 213.137.73.176 -> 192.203.175.9 SIP Status: 404 Not Found
 78.543317 213.137.73.176 -> 192.203.175.9 SIP Status: 488 Not Acceptable 
Media
 78.544346 192.203.175.9 -> 213.137.73.178 SIP Request: ACK 
sip:1401949nnnn at 213.137.73.178
 80.062729 213.137.73.176 -> 192.203.175.9 SIP Status: 488 Not Acceptable 
Media
 83.111677 213.137.73.176 -> 192.203.175.9 SIP Status: 488 Not Acceptable 
Media
 88.659719 213.137.73.176 -> 192.203.175.9 SIP Status: 404 Not Found


Another example...

85.770587 192.203.175.9 -> 213.137.73.178 SIP/SDP Request: INVITE 
sip:14014804420 at 213.137.73.178, with session description
 85.897768 213.137.73.176 -> 192.203.175.9 SIP Status: 100 Trying
 85.898134 213.137.73.176 -> 192.203.175.9 SIP Status: 407 Proxy 
Authentication Required
 85.898581 192.203.175.9 -> 213.137.73.178 SIP Request: ACK 
sip:1401480nnnn at 213.137.73.178
 85.899266 192.203.175.9 -> 213.137.73.178 IP Fragmented IP protocol 
(proto=UDP 0x11, off=552)
 85.899294 192.203.175.9 -> 213.137.73.178 SIP Request: INVITE 
sip:1401480nnnn at 213.137.73.178
 86.097458 213.137.73.176 -> 192.203.175.9 SIP Status: 100 Trying
 89.116528 213.137.73.176 -> 192.203.175.9 SIP Status: 180 Ringing
100.312843 213.137.73.176 -> 192.203.175.9 SIP Status: 488 Not Acceptable 
Media
100.313648 192.203.175.9 -> 213.137.73.178 SIP Request: ACK 
sip:1401480nnnn at 213.137.73.178
101.812307 213.137.73.176 -> 192.203.175.9 SIP Status: 488 Not Acceptable 
Media
104.851267 213.137.73.176 -> 192.203.175.9 SIP Status: 488 Not Acceptable 
Media
110.899191 213.137.73.176 -> 192.203.175.9 SIP Status: 488 Not Acceptable 
Media



--On Tuesday, March 11, 2003 8:04 PM -0600 Mark Spencer 
<markster at digium.com> wrote:

> Actually I think it was an issue with incrementing the sequence number on
> the bye andshould be fixed now.  RTCP is irrelevant in SIP signalling.
>
> Mark
>
> On Tue, 11 Mar 2003 alex at pilosoft.com wrote:
>
>> > 1 - From watching the udp fly by, it seems that iconnect does not know
>> > when we hang up.  For example, if I call a voice mail and it starts
>> > giving me its speal and I hang up, iconnect stays connected until the
>> > VM hangs up at its end.
>> Because Asterisk doesn't implement RTCP.