Our servers also use double Record-Route headers and we have always used SIPp in our testing with no issues. There are no inherent faults in the most recent version of SIPp with Record-Route/Route handling as far as I know.
As long as you are properly setting “rrs=true” on the received INVITE, and including the “<routes>” variable in your replies it all works perfectly. https://sipp.sourceforge.net/doc/reference.html#Actions Ben Newlin From: Users <[email protected]> on behalf of Thomas Pircher via Users <[email protected]> Date: Thursday, October 13, 2022 at 4:26 AM To: [email protected] <[email protected]> Cc: John Quick <[email protected]> Subject: Re: [OpenSIPS-Users] Problem proxying a SIP connection with t_relay EXTERNAL EMAIL - Please use caution with links and attachments John Quick wrote: >The UAS at 10.30.9.11 has failed to process the two Record-Route headers >sent in the INVITE. It should send the Route Set back as part of the >Response - i.e. within the 200 OK. But it hasn't. It has just absorbed the >Record-Route headers and ignored them. I would say that is faulty UAS >behaviour, but maybe Bogdan could confirm. Hi John, thanks for the reply. Your explanation makes sense to me; I can see that in the packet capture file, in the replies from the UAS in packets 4 and 6. Also, your article explains why OpenSIPS adds two RR headers in this scenario. >Consequently, the ACK has no Route headers. That means OpenSIPS is treated >as the final destination - it doesn't know that it is meant to relay the ACK >to 10.30.9.11 Now I have the right keywords to search for some more information; it looks like there was an attempt to fix this in 2006: https://sourceforge.net/p/sipp/mailman/sipp-users/thread/200606071744.k57HiPJ4002550%40mail.zserv.tuwien.ac.at/#msg9012298 But then there is http://yuminstallgit.blogspot.com/2011/03/record-route-and-route-fun-in-sipp.html and the comment from 2021 at the end suggests others have seen the same issue relatively recently. >If you can't fix the UAS, you could try using the Topology hiding module in >OpenSIPS. That would probably overcome the problem because Topology hiding >doesn't send Record-Route headers downstream. That gives me a few options; I'll try replacing the SIPp UAS with FreeSWITCH. This may sound a bit over-engineered, as all I need is a machine that automatically answers calls to a bunch of usernames and plays an audio file. But it gives me a scenario that vaguely resembles a real-world setup, to test against. Thanks, Thomas _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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