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

[Clif Jones]

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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From: Clif Jones <ctjones at earthlink.net>
Subject: [Asterisk-Users] Having problems with RTP packets and Hold
Date: Tue, 10 Feb 2004 08:03:55 -0500
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