[SR-Users] One way audio on calls from pstn

[Dmitri Korotkov]

Solved.

in asterisk<->kamailio trunk config placed

nat=yes
canreinvite=no

Thank you for cooperation,
Dmitri

02.07.2010 0:51, dotnetdub пишет:
>
>
> On 1 July 2010 22:41, Dmitri Korotkov <dmitri.korotkov at festart.ee 
> <mailto:dmitri.korotkov at festart.ee>> wrote:
>
>     Hi,
>
>     voice:/# ps auxf |grep rtpproxy |grep -v grep
>     rtpproxy  1291  0.0  0.0  26800   876 ?        Ssl  Jun18   0:10
>     /usr/sbin/rtpproxy -u rtpproxy rtpproxy -l my.public.ip.here -s
>     udp:localhost 7722
>     voice:/#
>
>
>     kamailio.cfg:
>     #!define WITH_MYSQL
>     #!define WITH_AUTH
>     #!define WITH_ACCDB
>     #!define WITH_NAT
>     #!define WITH_PSTN
>
>     #!ifdef WITH_NAT
>     loadmodule "nathelper.so"
>     #!endif
>
>     # ----- nathelper -----
>     #!ifdef WITH_NAT
>     modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7722
>     <http://127.0.0.1:7722>")
>     modparam("nathelper", "natping_interval", 30)
>     modparam("nathelper", "ping_nated_only", 1)
>     modparam("nathelper", "sipping_bflag", 7)
>     modparam("nathelper", "sipping_from", "sip:pinger at kamailio.org"
>     <mailto:sip:pinger at kamailio.org>)
>     modparam("registrar|nathelper", "received_avp", "$avp(i:80)")
>     modparam("usrloc", "nat_bflag", 6)
>     #!endif
>
>
>
>     02.07.2010 0:32, dotnetdub пишет:
>>
>>
> I'm not overly familiar with rtpproxy as we use mediaproxy but you 
> will need to engage it somewhere in your script, are you doing that?
>
> Take a look at 
> http://www.voip-info.org/wiki/view/Kamailio+1.5.x+and+RTPProxy
>
> Can you see any rtpproxy messages in syslog?
>
>
>>     On 1 July 2010 21:53, Dmitri Korotkov <dmitri.korotkov at festart.ee
>>     <mailto:dmitri.korotkov at festart.ee>> wrote:
>>
>>         Hello,
>>
>>         I have kamailio installation WITH_PSTN, WITH_NAT and rtpproxy.
>>         Using following scenario:
>>         [kamailio]<-sip trunk ->[asterisk gw] <->sip trunk <-> [PSTN
>>         provider]
>>
>>         All kamailio sip subscribers are behind nat in different
>>         networks.
>>
>>         1. OK. Local kamailio users can call one to other even they
>>         are on different networks behind nat.
>>         2. OK. Outgoing calls from kamailio users to PSTN work also
>>         very well.
>>         3. Not OK.  Incoming from PSTN side calls have only one way
>>         audio.
>>
>>         I tcpdump'ed kamailio box and found, that pstn provider sends
>>         RTP packets to kamailio IP in case of answered call.
>>
>>         I guess that rtpproxy is not active in case of pstn call.  Is
>>         it true ?
>>
>>         I am using more less default kamailio config
>>
>>         Could you please suggest solution ?
>>
>>         BR,
>>         Dmitri
>>
>>
>>
>>     Hi Dmitri,
>>
>>     Check out the nathelper module.
>>
>>     Regards,
>>     Brian
>>
>
>

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