[Freeswitch-users] Different SIP profiles for different codecs?
DJB INTERNATIONAL
djbinter at gmail.com
Thu Apr 22 08:32:27 PDT 2010
<action application="bridge"
data="{absolute_codec_string=G729}sofia/external//$0@ <0 at 172.16.15.11>
xxx.xxx.xxx.xxx"/>
-djbinter
On Thu, Apr 22, 2010 at 8:14 AM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:
> I need the internal codec to be hardcoded to PCMU. I need the PSTN codec to
> be 729, and I need to transcode between the two.
>
> So, we have PBX (711) ---> FS (729)---> PSTN and vice versa.
>
> If I only set the following parameters in one profile, I won't get the
> necessary result:
>
> <param name="inbound-codec-prefs" value="PCMU"/>
> <param name="outbound-codec-prefs" value="G729"/>
>
> We are routing calls through the system like a softswitch / B2BUA / SBC. We
> don't have phones registering to the FS box like a PBX. We are using sipX
> for that.
>
> So, we should be able to use two profiles and set the codecs differently in
> each. This is show in the following example on the dial plan page:
>
> Example 5
>
> In this example we will demonstrate the use of profiles when using a
> FreeSWITCH endpoint that supports profiles, like mod_sofia. Assuming that
> we
> want to use different call settings (codecs, DTMF modes, etc) for sending
> the calls to different IP addresses, we can create different profiles. For
> example, in the configuration of sofia.conf, we see an example profile
> named
> "test", which we rename to profile1 for this example, and add a profile2
> for
> comparison:
>
> <profile name="profile1">
> <param name="debug" value="1"/>
> <param name="rfc2833-pt" value="101"/>
> <param name="sip-port" value="5060"/>
> <param name="dialplan" value="XML"/>
> <param name="dtmf-duration" value="100"/>
> <param name="codec-prefs" value="PCMU at 20i"/>
> <param name="codec-ms" value="20"/>
> <param name="use-rtp-timer" value="true"/>
> </profile>
> <profile name="profile2">
> <param name="debug" value="1"/>
> <param name="rfc2833-pt" value="101"/>
> <param name="sip-port" value="5070"/>
> <param name="dialplan" value="XML"/>
> <param name="dtmf-duration" value="100"/>
> <param name="codec-prefs" value="PCMA at 20i"/>
> <param name="codec-ms" value="20"/>
> <param name="use-rtp-timer" value="true"/>
> </profile>
>
> The difference between the two profiles are in the codecs. The first uses
> G.711 uLaw and the second G711 ALaw.
>
> Continuing the examples above, we have:
>
> <extension name="Test5ulaw">
> <condition field="network_addr" expression="^192\.168\.1\.1$"/>
> <condition field="destination_number" expression="^1(\d+)$">
> <action application="bridge" data="sofia/profile1/$0 at 192.168.2.2"/>
> </condition>
> </extension>
>
> to send the call in G.711 uLaw and
>
> <extension name="Test5alaw">
> <condition field="network_addr" expression="^192\.168\.1\.1$"/>
> <condition field="destination_number" expression="^1(\d+)$">
> <action application="bridge" data="sofia/profile2/$0 at 192.168.2.2"/>
> </condition>
> </extension>
>
> But when we try to setup two profiles, we always get an error on one of
> them.
>
> Ken
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven
> Ayre
> Sent: Thursday, April 22, 2010 9:56 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Different SIP profiles for different
> codecs?
>
> No need to have separate profiles for it, these are the two parameters
> you'd want to change:
> <param name="inbound-codec-prefs" value="PCMU"/>
> <param name="outbound-codec-prefs" value="G729"/>
>
> If you want to have calls coming in one one profile (e.g. internal)
> and going out on another (e.g. external) you can do so. Create both
> profiles (if you haven't already) and set inbound-codec-prefs on
> internal and outbound-codec-prefs on external. Have the internal
> profile hit the dialplan context where you have the extensions
> configured and have the dialplan bridge to sofia/external/... to send
> the outgoing legs through that context.
>
> -Steve
>
>
> On 22 April 2010 15:30, Ken Fulmer <kenfulmer at icstechnologysolutions.com>
> wrote:
> > We are using the following external sip profile:
> >
> >
> >
> > <profile name="external">
> >
> > <settings>
> >
> > <param name="accept-blind-auth" value="true"/>
> >
> > <param name="apply-inbound-acl" value="domains"/>
> >
> > <param name="debug" value="0"/>
> >
> > <param name="sip-trace" value="no"/>
> >
> > <param name="rfc2833-pt" value="101"/>
> >
> > <param name="sip-port" value="5080"/>
> >
> > <param name="dialplan" value="XML"/>
> >
> > <param name="context" value="public"/>
> >
> > <param name="dtmf-duration" value="2000"/>
> >
> > <param name="inbound-codec-prefs" value="G729,PCMU"/>
> >
> > <param name="outbound-codec-prefs" value="G729,PCMU"/>
> >
> > <param name="hold-music" value="$${hold_music}"/>
> >
> > <param name="rtp-timer-name" value="soft"/>
> >
> > <param name="manage-presence" value="false"/>
> >
> > <param name="inbound-codec-negotiation" value="generous"/>
> >
> > <param name="nonce-ttl" value="60"/>
> >
> > <param name="auth-calls" value="false"/>
> >
> > <param name="rtp-ip" value="10.10.3.12"/>
> >
> > <param name="sip-ip" value="10.10.3.12"/>
> >
> > <param name="rtp-timeout-sec" value="300"/>
> >
> > <param name="rtp-hold-timeout-sec" value="1800"/>
> >
> > <param name="user-agent-string" value="INTEGRAL PARTNERS"/>
> >
> > </settings>
> >
> > </profile>
> >
> >
> >
> > The dial plan “public” context has two entries, one for PSTN access and
> one
> > for call routing to an internal PBX.
> >
> >
> >
> > This is the dial plan for PSTN call routing:
> >
> >
> >
> > <extension name="LCL">
> >
> > <condition field="destination_number" expression="^(\d{10})$">
> >
> > <action application="set" data="hangup_after_bridge=true"/>
> >
> > <action application="bridge"
> data="sofia/external/$1 at 172.16.15.11"/>
> >
> > <action application="set" data="ringback=${us-ring}"/>
> >
> > <action application="set" data="ignore_early_media=true" />
> >
> > </condition>
> >
> > </extension>
> >
> > <extension name="LD">
> >
> > <condition field="destination_number" expression="^1(\d{10})$">
> >
> > <action application="set" data="hangup_after_bridge=true"/>
> >
> > <action application="bridge"
> data="sofia/external/$0 at 172.16.15.11"/>
> >
> > <action application="set" data="ringback=${us-ring}"/>
> >
> > <action application="set" data="ignore_early_media=true" />
> >
> > </condition>
> >
> > </extension>
> >
> > <extension name="911">
> >
> > <condition field="destination_number" expression="^(911)$">
> >
> > <action application="set" data="hangup_after_bridge=true"/>
> >
> > <action application="bridge"
> data="sofia/external/$1 at 172.16.15.11"/>
> >
> > <action application="set" data="ringback=${us-ring}"/>
> >
> > <action application="set" data="ignore_early_media=true" />
> >
> > </condition>
> >
> > </extension>
> >
> > <extension name="INTL">
> >
> > <condition field="destination_number" expression="^011(\d+)$">
> >
> > <action application="set" data="hangup_after_bridge=true"/>
> >
> > <action application="bridge"
> data="sofia/external/$0 at 172.16.15.11"/>
> >
> > <action application="set" data="ringback=${us-ring}"/>
> >
> > <action application="set" data="ignore_early_media=true" />
> >
> > </condition>
> >
> > </extension>
> >
> >
> >
> > And this is the entry for internal call routing to a PBX:
> >
> >
> >
> > <extension name="PBX">
> >
> > <condition field="destination_number" expression="^(205314849[0-9])$">
> >
> > <action application="set" data="hangup_after_bridge=true"/>
> >
> > <action application="bridge"
> > data="sofia/external/$1 at 10.10.3.10"|data="sofia/external/$1 at 10.10.3.11
> "/>
> >
> > <action application="set" data="ringback=${us-ring}"/>
> >
> > <action application="set" data="ignore_early_media=true"/>
> >
> > </condition>
> >
> > </extension>
> >
> >
> >
> > So, here’s my question:
> >
> >
> >
> > We’d like to be able to lock down the codec as 711 for the internal leg
> > going to the PBX and 729 for the external leg to the PSTN. We have
> > transcoding setup and it’s working fine. How can we use two SIP profiles
> to
> > hard code the codec in each direction? I’ve seen an example in the dial
> plan
> > section, but didn’t understand how to implement it.
> >
> >
> >
> > Thanks,
> >
> >
> >
> > Ken Fulmer
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
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