[Fwd: [Asterisk-Users] Having problems with RTP packets and H old]

[Regovich, Timothy]

Why does the FXO gateway treat a lack of RTP packets as a dropped call (and
what heuristic does it use to determine?)
Until the SIP UA sends an actual BYE message, the Dialog should still be
considered active, regardless of the RTP that may or may not be happening.



-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Clif Jones
Sent: Tuesday, February 10, 2004 1:33 PM
To: asterisk-dev at lists.digium.com; asterisk users
Subject: [Fwd: [Asterisk-Users] Having problems with RTP packets and Hold]


If anyone is familiar with the SIP SDP handling routines I would appreciate
some

insight.  The following problem that I found using Asterisk appears to be
improper

handling of a call put on hold when there is no music on hold:

[FXO gateway]                         [Asterisk]
[IP phone]

    |-------[INVITE s/SDP]---------------->|-------[INVITE
s/SDP]---------------->|
    |                                      |
|
    |<--------[180 Ringing]----------------|<--------[180
Ringing]----------------|
    |                                      |
|
    |<----[183 Session Progress]-----------|<-----------[200
OK/SDP]--------------|
    |                                      |
|
    |<--------[200
OK/SDP]-----------------|------------[ACK]-------------------->|
    |                                      |<=========== RTP
====================>|
    |------------[ACK]-------------------->|
|
    |<=========== RTP ====================>|
|

                                                 {IP phone puts caller on
hold}

    |                                      |<-----[INVITE/held
SDP]---------------|
    |                                      |
|
    |                                      |-----------[200
OK/SDP]-------------->|
    |                                      |
|
    |
|<------------[ACK]--------------------|
    |============ RTP (one-way)===========>|
|
    |                                      |
|
    |----------[BYE]---------------------->|
|
    |                                      |
|
    |<------------[200 OK]-----------------|
|

When the IP phone puts the gateway on hold, Asterisk gets the re-INVITE with
held
media but Asterisk doesn't re-INVITE the gateway.  The RTP traffic to the
gateway
stops so the gateway handles the condition as a lost connection.  Shouldn't
asterisk
be forwarding the re-INVITE to the gateway unless MOH is enabled?




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