[Freeswitch-users] Amazon EC2 no audio

[Larry Edelstein]

I had this same problem but eventually overcame it.  I modified the docs at

http://wiki.freeswitch.org/wiki/Amazon_ec2

accordingly.  I think the problem I had was the internal vs. external IP
address, as you've alluded to at the bottom of your message.

In addition to the changes you've made, I also modified
sip_profiles/internal.xml - see the docs.

-larry


On Wed, May 6, 2009 at 9:20 PM, Dave Grootwassink <stackofstuff.dg at gmail.com
> wrote:

>  Hello all,
>
>
>
> Help a n00b out.    I have been trying to get an instance of FreeSwitch
> running up in the Amazon EC2 cloud.
>
>
>
> I have successfully gotten the package built following the wiki and
> archives of this list.
>
>
>
>  I can get x-lite to register with the switch and it will set up calls out
> on my asterlink account.  The problem is that there is no audio transfer (so
> I am assuming RTP problem).
>
>
>
> The setup:
>
>
>
> Firewall open ports
>
> tcp 0-65535
>
> udp 0-65535        --- I tried so many combinations unsuccessfully, I
> finally just blasted open everything.
>
>
>
>
>
> In conf/freeswitch.xml   (174.129.201.96  is assigned elastic IP address)
>
>
>
>   <X-PRE-PROCESS cmd="include" data="vars.xml"/>
>
>   #set "external_rtp_ip=174.129.201.96"
>
>   <section name="configuration" description="Various Configuration">
>
>     <X-PRE-PROCESS cmd="include" data="autoload_configs/*.xml"/>
>
>   </section>
>
>
>
>
>
> In conf/autoload_configs/sofia.conf.xml
>
> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>
> <param name="ext-sip-ip" value="$${external_rtp_ip}"/>
>
>
>
>
>
> Internal network IP assignment:
>
> Name:   domU-12-31-39-00-84-B6.compute-1.internal
>
> Address: 10.254.139.68
>
>
>
>
>
>
>
> When I setup a call through asterlink, I see this in the system log:
>
>
>
> Ring SDP:
>
> v=0
>
> o=FreeSWITCH 1241648830 1241648831 IN IP4 10.254.139.68
>
> s=FreeSWITCH
>
> c=IN IP4 10.254.139.68
>
> t=0 0
>
> m=audio 17654 RTP/AVP 0 101
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:101 telephone-event/8000
>
> a=fmtp:101 0-16
>
> a=silenceSupp:off - - - -
>
> a=ptime:20
>
> a=sendrecv
>
>
>
> I do not know SDP parameters off the top of my head, but I am assuming that
> it is telling Asterlink to route the RTP to the internal network IP address
> and not the external one.
>
>
>
>
>
>
>
>
>
>
>
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