Subject: [Sip] Re-invite Without SDP

From: Nitin Kapoor <[email-masked]>
Date: Fri, 27 Nov 2009 16:06:15 -0500

Hello All,

I am facing the issue with one of my customer at the time of unhold the
call.

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.

Thanks,
Nitin Kapoor
General Telecom

---

From: "Dale Worley" <[email-masked]>
Date: Sat, 28 Nov 2009 01:29:06 -0500

On Fri, 2009-11-27 at 16:06 -0500, Nitin Kapoor wrote:
> sending the another "invite without sdp" than my SBC/Nextone is
> sending "488 unacceptable" to my source UA.

If the SBC won't accept an INVITE without SDP, it is operating
incorrectly and you won't be able to get SIP systems to work with it.

Dale

---

From: [email-masked]
Date: Mon, 30 Nov 2009 15:18:41 +0800

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.

Best regards
Gerald.yan

Nitin Kapoor <[email-masked]> 
发件人:  [email-masked]
2009-11-28 05:06

收件人
[email-masked]
抄送

主题
[Sip] Re-invite Without SDP

Hello All,

I have configured an endpoint that is dropping calls when you place the 
call on hold.

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.

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. 

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)

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.

Thanks,
Nitin Kapoor
General Telecom

[附件 "Bad Call.pcap" 被 严成安170027/user/zte_ltd 删除][附件 "Good 
Call.pcap" 被 严成安170027/user/zte_ltd 删除]
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