[OpenSER-Users] Missing RTP stream

Morten Isaksen misak at misak.dk
Thu Sep 20 23:59:32 CEST 2007


Hi!

I can see in the mediaproxy log the it is initialized to proxy the
call, but I newer get a "session xxxxx: called signed in from xxx"
from Asterisk.

session.py shows that the the connection between mediaproxy and
Asterisk is missing.

I will try to take a look at the sip debug from asterisk and try to
change the NAT settings in Asterisk.

Thanks for your input.

On 9/20/07, Norman Brandinger <norm at goes.com> wrote:
> You stated that you've forced every call through mediaproxy.  Are you
> positive ?
>
> Have you taken a look at the mediaproxy logs (and/or sessions.py  when
> the call is up) ?  They might provide some useful information to you.
>
> Ditto for Asterisk "sip set debug on" (note that the sip debug command
> format is a moving target).
>
> Have you looked at the "nat=" settings in sip.conf as well ?  At times,
> they tie closely with "canreinvite=".
>
> Norm
>
>
> Morten Isaksen wrote:
> > Hi!
> >
> > canreinvite is set to no and the OpenSER/mediaproxy is "hiding" the
> > clients IP-addresses from Asterisk, so I am pretty sure that this is
> > not the issue.
> >
> > On 9/20/07, Norman Brandinger <norm at goes.com> wrote:
> >
> >> Hi Morten,
> >>
> >> Admittedly, I haven't looked closely at your trace.  However, based on
> >> the description you gave, the first place to look is at the "canrevite"
> >> setting in Asterisk sip.conf.  You might want to try "canreinvite=no"
> >> after reading up on this particular setting.
> >>
> >> Regards,
> >> Norm
> >>
> >>
> >> Morten Isaksen wrote:
> >>
> >>> Hi!
> >>>
> >>> I have a strange problem with a missing RTP stream between OpenSER and
> >>> Asterisk. I am not sure if it is OpenSER og Asterisk related.
> >>>
> >>> I have this setup
> >>>
> >>> Phone A (172.17.96.17) --
> >>>                                       \      Openser    --    Asterisk
> >>>       --    PSTN
> >>>                                       /      (192.168.0.6)   (192.168.0.3)
> >>> Phone B (172.17.96.10) --        (172.17.64.1)
> >>>
> >>> I also have a Mediaproxy running on OpenSER and I force every call to
> >>> use the Mediaproxy.
> >>>
> >>> I call from Phone A or B to the PSTN works fine and from PSTN to Phone
> >>> A or B it also works.
> >>>
> >>> I have the dialplan logic on my Asterisk server so I want calls from
> >>> Phone A to Phone B to pass the Asterisk server. And this is were I
> >>> have the problem. When the call is established the RTP stream is
> >>> missing between Mediaproxy and Asterisk. I only have a RTP stream
> >>> between the phones and Mediaproxy. As far as I can see the SIP
> >>> signalling is correct.
> >>>
> >>> The SIP traces is listed below. Can you spot the problem in this?
> >>>
> >>> I will buy a beer (or 5) at OpenSER training in Rome to anyone who can
> >>> help me solve this problem.
> >>>
> >>> SIP trace between the phones and OpenSER:
> >>>
> >>>
> >
> >
> >
>
>


-- 
Morten Isaksen
http://www.misak.dk/blog/




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