The SBC may not support the offer/answer
model 200 OK(SDP) for offer and ACK(SDP) for answer.
Then, you can try to use another method
to implement Call Hold service, besides using re-INVITE
without SDP for Unhold.
|Nitin Kapoor <nitinkapoorr <at> gmail.com>
发件人: sip-bounces <at> ietf.org
|sip <at> ietf.org
|[Sip] Re-invite Without SDP
I am facing the issue with one of my customer at the time of unhold the
I have configured an endpoint that is dropping calls when you place the
call on hold.
The network configuration is as follows:
Cisco CallManager using SIP signaling > Cisco Unified Border Element
(CUBE) using SIP signaling > MSX communicates SIP to CUBE and h.323
to HT_5850_Egress > PSTN
When a phone on the Cisco CallManager places a call to a user on the PSTN
the call goes through successfully. When the phone on the Cisco CallManager
places that call on hold, the call gets dropped.
The message I see in the SIP trace is '488 Not acceptable here'.
If we do this same test from the Call Manager to the CUBE to the MSX but
instead to L3_Egress so the signalling between the CUBE/MSX/PSTN is SIP
the hold option works.
First IWF.. Where this option is not working.
Calling number 763.432.8682
Called number 763.577.3948
On this First Source UA has send the initial invite and which SBC has sent
to termination END as h.323 and call is connected successully after the
the mesages... Now when my Source UA has sent the invite to put that call
on hold it sends "re-invite with SDP with codec g.711u and media
attribute = inactive", and call goes on hold successfully, but
now whenever they are trying to unhold the call and sending the another
"invite without sdp" than my SBC/Nextone is sending "488
unacceptable" to my source UA.
And now the other option SIP(both legs)
Calling number 763.432.8682
Called number 1612.964.8862
Now in this scenario i have both the legs on sip and when my source UA
is sending the "invite without SDP" to unhold this call then
my my termination end is sending 200 OK with SDP to ACK that call and unhold
scenario is working fine.
I am also attaching the call flow for both the scenario. Please help me
to find out the root cause of this issue.
[附件 "Bad Call.pcap" 被 严成安170027/user/zte_ltd 删除][附件
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