Re: [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.korot...@festart.ee <mailto:dmitri.korot...@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:pin...@kamailio.org"
    <mailto:sip:pin...@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.korot...@festart.ee
    <mailto:dmitri.korot...@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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