[Freeswitch-users] RTP reception before sending any RTP packet

[Mickael Hubert]

Thanks for your answer Vallimamod,

You can see our dialplan bellow:
We tried these commands "rtp auto adjust, etc ...", but the issue continues.

is there a way to show if FS knows when the call is natted or not ? FYI we
have an opensips of front for SIP, but RTP is send directly to FS.

SIP call --> Opensips (nat detection) --> FS
RTP flow ------------------------------------------> FS

<condition field="${destinationAuthorized}" expression="^yes$"
break="never">
         <action application="export"
data="codec_string=${ep_codec_string}"/>
         <action application="export" data="sip_h_X-cid=${uuid}"/>
         <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="ignore_early_media=false"/>
          <action application="set" data="ignore_display_updates=true"/>
          <action application="set" data="inherit_codec=true"/>
          <action application="set" data="fax_enable_t38=false"/>
          <action application="export" data="suppress_cng=true"/>
          <action application="set" data="disable_rtp_auto_adjust=false"/>
          <action application="set" data="rtp_auto_adjust_threshold=1"/>
          <action application="set"
data="rtp_manual_rtp_bugs=accept_any_packets"/>
          <action application="unset" data="sip_h_X-Username"/>
          <action application="bridge" data="
          {
              sip_invite_params=user=phone,
              sip_cid_type=none,
              ignore_display_updates=true,
               sip_contact_user=${caller_id_number}
          }
          sofia/internal/${destination_number}@${distributor(LISTINTERNAL
${sofia(profile internal gwlist down)})}"/>

Thanks in advance


2018-04-19 11:35 GMT+02:00 Vallimamod Abdullah <vma at vallimamod.org>:

> Hi,
>
> To my knowledge, it is the default beahviour for freeswitch to wait for
> incoming rtp from nated endpoints before starting to transmit.
>
> You can explicitely force it with the profile param
> disable-rtp-auto-adjust=false.
> You can also play with the channel variable rtp_auto_adjust_threshold
> which defines the number of rtp packets to wait for initially (default is
> 10 iirc)
>
> Best Regards,
> --
> Vallimamod Abdullah
> SIP Solutions
> linkedin.com/in/vallimamod
> .
>
> > On 19 Apr 2018, at 10:31, Mickael Hubert <mickael at winlux.fr> wrote:
> >
> > Hi list,
> > I'd like to know if there's a way to tell FS to wait for RTP reception
> before sending any RTP packet ?
> > We're facing a problem regarding early media with a PBX, NATed behind a
> dumb router : actually, if FS first sends RTP to the PBX, without waiting
> for RTP coming from the PBX, there's no audio. In some cases (related to
> the provider we use behind FS to establish call to the PSTN) FS isn't the
> 'first-shooter' of the RTP flow, and everything works fine, but when FS is
> the first to send RTP, it doesn't work...
> > Because we can't change anything at our providers configs, we'd like to
> know if FS can "delay" RTP sending until it receives some RTP from the
> calling party.
> >
> > The schematic is following :
> > PBX ===> FS ===> ...voice backbone... ===> Providers
> > (FS is acting as an SBC)
> >
> > Thanks for your help,
> >
>
>
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