[Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch issue

Mon Dec 19 15:29:44 MSK 2011

Hi,
I'm having hard time understanding which one is FreeSWICTH and which on is
openSIPS. Looking at the sip tarces however, I must say you should at-max
have a one-way audio because the remote end-point is sending its Public
Address for Media-connectivity, you freeswitch in return is sending its
private IP to OpenSIPS and openSIPs is relaying the same Private-IP to the
remote-end point...

U 192.168.23.34:5060 -> 184.150.225.230:5062
SIP/2.0 200 OK.
Via: SIP/2.0/UDP
184.150.225.230:5062;received=184.150.225.230;rport=5062;branch=z9hG4bK-4008200999-3775991592-3239888547-3704563422.
Record-Route: <sip:192.168.23.34;lr;ftag=917850919-3775991592-3239888547-3704563422;did=5df.895f858>.
From: <sip:14167125040 at 184.150.225.230:5062;user=phone>;tag=917850919-3775991592-3239888547-3704563422.
To: <sip:16474272135 at 66.240.179.221;user=phone>;tag=9mr8cyc2pgjmS.
Call-ID: 274bb6c6280f11e1a3c61cc1de26cfdc at 184.150.225.230.
CSeq: 1 INVITE.
Contact: <sip:16474272135 at 192.168.23.33:5060;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f4320b5 2011-11-28 08-27-46 -0600.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
sla, include-session-description, presence.winfo, message-summary,
refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 249.
Remote-Party-ID: "16474272135"
<sip:16474272135 at 66.240.179.221>;party=calling;privacy=off;screen=no.
.
v=0.
o=FreeSWITCH 1324008356 1324008357 IN IP4 *192.168.23.33*.
s=FreeSWITCH.
c=IN IP4 *192.168.23.33*.
t=0 0.
m=audio 49774 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.

And you know the remote end-point is sending the media to provate ip of FS
within its LAN...!!

So, Like I said use RTPproxy module such that newer INVITE look Like below..

U 192.168.23.34:5060 -> 184.150.225.230:5062
SIP/2.0 200 OK.
Via: SIP/2.0/UDP
184.150.225.230:5062;received=184.150.225.230;rport=5062;branch=z9hG4bK-4008200999-3775991592-3239888547-3704563422.
Record-Route: <sip:192.168.23.34;lr;ftag=917850919-3775991592-3239888547-3704563422;did=5df.895f858>.
From: <sip:14167125040 at 184.150.225.230:5062;user=phone>;tag=917850919-3775991592-3239888547-3704563422.
To: <sip:16474272135 at 66.240.179.221;user=phone>;tag=9mr8cyc2pgjmS.
Call-ID: 274bb6c6280f11e1a3c61cc1de26cfdc at 184.150.225.230.
CSeq: 1 INVITE.
Contact: <sip:16474272135 at 192.168.23.33:5060;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f4320b5 2011-11-28 08-27-46 -0600.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
sla, include-session-description, presence.winfo, message-summary,
refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 249.
Remote-Party-ID: "16474272135"
<sip:16474272135 at 66.240.179.221>;party=calling;privacy=off;screen=no.
.
v=0.
o=FreeSWITCH 1324008356 1324008357 IN IP4 *PU.BL.IC.IP of OpenSIPS*.
s=FreeSWITCH.
c=IN IP4 *PU.BL.IC.IP of OpenSIPS*
.
t=0 0.
m=audio 49774 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.

Wish you best of Luck.!

Regards,
Sammy

On Mon, Dec 19, 2011 at 5:12 PM, Peter Spinato <peter at spinato.ca> wrote:

> ** **
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus
> *Sent:* Monday, December 19, 2011 1:57 AM
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch
> issue****
>
> ** **
>
> If that doesn't help, can you provide an actual siptrace of the call so we
> can SEE what's going on?****
>
> ngrep can be quite helpful..****
>
>
> ****
>
> -Avi****
>
> ** **
>
> On Mon, Dec 19, 2011 at 6:41 AM, Sammy Govind <govoiper at gmail.com> wrote:*
> ***
>
> Hey, ****
>
> ** **
>
> I think you need to use RTP proxy in bridged mode on OpenSIPS and use the
> force_rtp_proxy() function with IE and EI flags so that the SDPs IPs change
> as follows****
>
> ** **
>
> End-USER<===========>[*Public-IP*]OPENSIPS/RTPProxy[*Private-IP*
> ]<===========>[*Private-IP*]FreeSWITCH****
>
> ** **
>
> Use FreeSWITCH internal or external domain, make sure you've correct
> properties and contexts set.****
>
> ** **
>
> Regards,****
>
> Sammy.****
>
> ** **
>
> On Mon, Dec 19, 2011 at 5:28 AM, Peter Spinato <peter at spinato.ca> wrote:**
> **
>
> I changed the external profile to port 5060 – I can see it using that
> profile and it still sends out the local IP at the RTP IP … even though
> ext-rtp-ip is set to the public one …?****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus
> *Sent:* Saturday, December 17, 2011 10:32 PM****
>
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch
> issue****
>
>  ****
>
> Yes, send it to the freeswitch server port  5080****
>
> Or if you don't need the internal profile, remove it and set external to
> use port 5060.
> ****
>
> -Avi****
>
>  ****
>
> On Sun, Dec 18, 2011 at 5:07 AM, Peter Spinato <peter at spinato.ca> wrote:**
> **
>
> The calls go to internal profile – I guess case the OpenSIPs server is
> local lan – is there a way to force that calls to external profile as I
> find calls that hit that profile load the ext-rtp-ip ip and work.****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus
> *Sent:* Saturday, December 17, 2011 11:07 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch
> issue****
>
>  ****
>
> Someone familiar with this might be able to answer right off the bat, but
> if you pastebin a SIP trace (and an fs_cli log for completeness) , the
> problem should become apparent.****
>
>  ****
>
> Are you using the internal profile for all the calls? If you use the
> external.xml profile, you might need to set your ext-rtp-ip in that file,
> too. Seeing a trace will tell (almost) the whole story.****
>
>
> ****
>
> -Avi****
>
>  ****
>
> On Fri, Dec 16, 2011 at 10:46 PM, Peter Spinato <peter at spinato.ca> wrote:*
> ***
>
> All,****
>
>   Hopefully someone can assist me - I'll gladly give $50 to the person who
> helps me fix the issue - I have an OpenSIPs server configured as a load
> balancer (http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS)
> that receives the call and forwards it to my Freeswitch Server for an IVR.
> When I had the call just routing to the Freeswitch server I got the audio
> working by setting the ext-rtp-ip to the public IP.  Now that I route the
> SIP call through the OpenSIPs server there is no audio - I'm guessing it a
> NAT issue as always.****
>
>  ****
>
> Both the OpenSIPS and FreeSwitch server have an internal private IP -
> OpenSIPS = 192.168.23.1 and FreeSwitch 192.168.23.2****
>
> Both servers also have a public IP that routes to the internal IP OpensIPS
> = 47.1.1.1 and Freeswitch = 47.1.1.2****
>
>  ****
>
> Call gets received by the OpenSIPs via the external IP - routes the call
> to the internal IP on the Freeswitch server which answers the call - but no
> audio - I think the FreeSwitch is trying to route the RTP Audio via its
> internal private IP instead of the public IP of 47.1.1.2.  Not sure if this
> is the real issue or how to configure it route RTP properly  ... all help
> is appreciated.  $50 Paypal to whoever fixes this for me!  Thanks****
>
>  ****
>
> -Peter****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
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>  ****
> ------------------------------
>
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2012.0.1890 / Virus Database: 2108/4686 - Release Date: 12/17/11*
> ***
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
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>
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> ------------------------------
>
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> Checked by AVG - www.avg.com
> Version: 2012.0.1890 / Virus Database: 2108/4686 - Release Date: 12/17/11*
> ***
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
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>
> ** **
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
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> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
> ** **
> ------------------------------
>
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2012.0.1890 / Virus Database: 2108/4689 - Release Date: 12/18/11*
> ***
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
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