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issued to Asterisk with an inactive SDP attribute. Asterisk replies with
inactive SDP and re-INVITES the SIP PSTN channel so the RTP is now going
through Asterisk, music on hold is played to the PSTN channel.

Cisco then re-INVITES with no SPD, Asterisk responds with sendrecv and
Cisco acknowledges with  sendonly SDP attribute, after this Cisco sends
music on hold directly to the SIP PSTN channel. Asterisk repeats the
previous re-INVITE on the PSTN channel and sends music on hold. The result
is that the held party receives two RTP streams, one from Cisco UCM and one
from Asterisk, each playing music on hold.

It is believed that Cisco issues inactive between placing the call on hold
and generating music on hold. This also appears to be performed when the
call is taken off hold. 

It is believed that when Asterisk receives the sendonly it should stop
sending RTP to the held channel. 

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 (0114645) davidw (reporter) - 2009-12-03 05:09
 https://issues.asterisk.org/view.php?id=16373#c114645 
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By using a Polycom, rather than the Cisco, one can also demonstrate this,
in terms of dialogues, and final RTP configuration, without the
complications due to the Cisco sending re-invite with no SDP, and
temporarily re-inviting to a=inactive.

The big difference is that the Polycom doesn't source MOH, so the result
is not, or at least less, obvious to the user.

As phones send a=sendonly, without sourcing MOH, the best approach may be
to respond a=inactive, rather than a=recvonly.  Alternatively one could
re-invite the holder back to Asterisk, so that Asterisk had full control of
the RTP stream, although I don't know what happens if there is a conflict
between upstream RTP and local MOH, in Asterisk itself. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-12-03 05:09 davidw         Note Added: 0114645                          
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