Clif Jones
2004-Feb-10 11:33 UTC
[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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Subject: [Asterisk-Users] Having problems with RTP packets and Hold
Date: Tue, 10 Feb 2004 08:03:55 -0500
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