[OpenSER-Users] Missing RTP stream

[Morten Isaksen]

The error was in Mediaproxy.

In rtphandler.py i changed

nonPublicNetworks = [
    {'name': '0.0.0.0',     'value': 0x00000000L, 'mask': 0xff000000L},
    {'name': '10.0.0.0',    'value': 0x0a000000L, 'mask': 0xff000000L},
    {'name': '127.0.0.0',   'value': 0x7f000000L, 'mask': 0xff000000L},
    {'name': '172.16.0.0',  'value': 0xac100000L, 'mask': 0xfff00000L},
    {'name': '192.168.0.0', 'value': 0xc0a80000L, 'mask': 0xffff0000L},
    {'name': '224.0.0.0',   'value': 0xe0000000L, 'mask': 0xf0000000L}
]

To

nonPublicNetworks = [
    {'name': '0.0.0.0',     'value': 0x00000000L, 'mask': 0xff000000L},
    {'name': '10.0.0.0',    'value': 0x0a000000L, 'mask': 0xff000000L},
    {'name': '127.0.0.0',   'value': 0x7f000000L, 'mask': 0xff000000L},
    {'name': '224.0.0.0',   'value': 0xe0000000L, 'mask': 0xf0000000L}
]

I think Mediaproxy got confused with the RFC1918 IP's. In my setup
there is no NAT between 172.17.0.0/16 and 192.168.0.0/24 - just a
router.


On 9/21/07, Norman Brandinger <norm at goes.com> wrote:
> Is there a firewall in the picture ?  You have two different subnets and
> there probably is a box doing some (NAT) translation  / routing between
> them.  Is is possible the RTP stream is being blocked at the firewall ?
>
> Norm
>
>
> Morten Isaksen wrote:
> > 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/