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New Member

SIP call setup

I am testing 2 ata 188's using SIP. I am experiencing a situation where after 20 seconds the called phone hangs up. It appears to be a problem with the final ack during call setup.

Shoud the final ack in this call setup be sent from ata1 directly to ata2 or through the sip server.

Here is what I see via the call tree.

ata1 SipServer ata2


Trying <---------------

Invite ----------------->

Trying <---------------

Ringing <-----------------

Ringing <--------------

OK <---------------

OK <------------------

Ack ------------->

According to the RFC it looks like after the OK has been received by ata1 then ata1 is supposed to send the ACK directly to ata2. However it is sending it to the sip server. This particular sip server is dropping the ack, thus causing ata2 to resend the OK and wait for the ACK again. It never receives the ACK and after 20 seconds hangs up.

Am I understanding the RFC correctly, should the ata1 send the ack to ata2 directly.

New Member

Re: SIP call setup

Rod, you are right about the RFC.

The ACK has to be sent directly to the remote UA. Based on that, the UA ATA1 could be at fault here (or not compliant with the SIP specs). You might have to contact Cisco’s TAC to understand their implementation of that particular piece of signal.

On the other hand, most SIP Proxies/Servers I have seen out there will go the extra mile and forward the ACK if the UA does send it to them. I will be very interested in knowing which SIP server (brand name/ model/version) you are using that is not forwarding the ACK. Another reason, which I know could cause the SIP server not to forward a particular signal, is it being erroneous. If a SIP server gets a badly constructed SIP packet, it could well drop it and give no answer back, in addition to not forwarding it.

On some UAs, you can turn on the “record route” feature, which will force the SIP server into processing the ACK and taking some type of action. I just am not familiar enough with the ATA to tell you whether this feature is supported or not.

I might have left you with more questions than asnwers, but i will appreciate a feedback from you regarding the identity of the SIP server you are using and any other solutions you come up with on this issue.



New Member

Re: SIP call setup

Thanks for the reply. I am using a download shareware of partysip version 0.5.0. That developer mentioned that the sip server would forward the ACK if record route is turned on. However I still haven't gotten that to work. Will try a newer versio of partysip. He also mentioned that the ata should be sending it directly so I will have to open up a case with CISCO on this unless someone else comes up with a fix before-hand.


Rod Frost

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