<div dir="ltr"><div><div>Hi Matt,<br><br></div>I also tested the directmedia=yes over 3g connection ie with a public ip but I am getting only one way audio<br></div>am I doing anything wrong?<br></div><div class="gmail_extra">
<br><br><div class="gmail_quote">On Wed, Jul 9, 2014 at 6:54 PM, Sameer Rathod <span dir="ltr"><<a href="mailto:sameer@hostnsoft.com" target="_blank">sameer@hostnsoft.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr"><div><div><div><div><div><div>Hi Matt,<br><br></div>Thank you so much for explaining me this concept <br><br></div>One more thing when I did testing for the above in different cases ie with directmedia=yes and no I got the flow of packets attached with this mail <br>
</div>Please have a look<br><br></div>The flow stats that the rtp packet flows directly between end point <br></div>So as per above details probably it is due to both of my endpoints are on the same network ie one side of the nat <br>
<br></div>am i right?<br><div><div><br><br><div><div><br><div><div><br></div></div></div></div></div></div></div><div class="gmail_extra"><div><div class="h5"><br><br><div class="gmail_quote">On Wed, Jul 9, 2014 at 6:36 PM, Matthew Jordan <span dir="ltr"><<a href="mailto:mjordan@digium.com" target="_blank">mjordan@digium.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div>On Wed, Jul 9, 2014 at 4:56 AM, Sameer Rathod <<a href="mailto:sameer@hostnsoft.com" target="_blank">sameer@hostnsoft.com</a>> wrote:<br>
> Hi,<br>
><br>
> with canreinvite=no and directmedia=no I and getting the message in the logs<br>
> for all calls<br>
><br>
> "switching from simple_bridge technology to native_rtp"<br>
><br>
><br>
> -- Executing [102@mkg:1] Dial("SIP/101-00000017", "SIP/102") in new stack<br>
> == Using SIP RTP CoS mark 5<br>
> -- Called SIP/102<br>
> -- SIP/102-00000018 is ringing<br>
> -- SIP/102-00000018 answered SIP/101-00000017<br>
> -- Channel SIP/101-00000017 joined 'simple_bridge' basic-bridge<br>
> <0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab><br>
> -- Channel SIP/102-00000018 joined 'simple_bridge' basic-bridge<br>
> <0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab><br>
> > Bridge 0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab: switching from<br>
> simple_bridge technology to native_rtp<br>
> > 0x7f427c068a10 -- Probation passed - setting RTP source address to<br>
> <a href="http://111.118.250.236:49344" target="_blank">111.118.250.236:49344</a><br>
> > 0x7f427c068a10 -- Probation passed - setting RTP source address to<br>
> <a href="http://111.118.250.236:49344" target="_blank">111.118.250.236:49344</a><br>
> > 0x7f42500168d0 -- Probation passed - setting RTP source address to<br>
> <a href="http://111.118.250.236:26326" target="_blank">111.118.250.236:26326</a><br>
> > 0x7f42500168d0 -- Probation passed - setting RTP source address to<br>
> <a href="http://111.118.250.236:26326" target="_blank">111.118.250.236:26326</a><br>
> -- Channel SIP/101-00000017 left 'native_rtp' basic-bridge<br>
> <0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab><br>
> -- Channel SIP/102-00000018 left 'native_rtp' basic-bridge<br>
> <0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab><br>
> == Spawn extension (mkg, 102, 1) exited non-zero on 'SIP/101-00000017'<br>
><br>
><br>
><br>
> I cannot understand why asterisk state diff bridges if all works same<br>
><br>
> please can anyone explain me the working bridging concept and how to<br>
> configure and use bridges to route the rtp externally form asterisk.<br>
><br>
<br>
</div></div>I think I just answered this in your other thread, but I'll repeat it here.<br>
<br>
First, canreinvite has been deprecated as a naming convention for ...<br>
a long time. It's not even documented any more. The code will accept<br>
it, but all you're doing is setting the directmedia option twice:<br>
<br>
} else if (!strcasecmp(v->name, "directmedia") ||<br>
!strcasecmp(v->name, "canreinvite")) {<br>
ast_set_flag(&mask[0], SIP_REINVITE);<br>
ast_clear_flag(&flags[0], SIP_REINVITE);<br>
<br>
The native RTP bridge in Asterisk 12 manages bridges between two RTP<br>
capable channels. The bridge can either be formed remotely (in which<br>
case the media flows between the endpoints) or locally, in which case<br>
the media is swapped across the ports. It will attempt to perform a<br>
remote bridge if possible, while falling back to a local bridge if a<br>
remote bridge is not possible.<br>
<br>
In your particular case, you've explicitly told it to *not* do<br>
directmedia. So it won't perform a remote bridge.<br>
<br>
Even if you set directmedia=yes (or one of its variants), you may not<br>
have a successful remote bridge if one of the endpoints is behind a<br>
NAT. The sip.conf sample configuration documentation is actually quite<br>
good on this subject:<br>
<br>
;----------------------------------- MEDIA HANDLING<br>
--------------------------------<br>
; By default, Asterisk tries to re-invite media streams to an optimal<br>
path. If there's<br>
; no reason for Asterisk to stay in the media path, the media will be<br>
redirected.<br>
; This does not really work well in the case where Asterisk is outside and the<br>
; clients are on the inside of a NAT. In that case, you want to set<br>
directmedia=nonat.<br>
;<br>
;directmedia=yes ; Asterisk by default tries to redirect the<br>
; RTP media stream to go directly from<br>
; the caller to the callee. Some devices do not<br>
; support this (especially if one of<br>
them is behind a NAT).<br>
; The default setting is YES. If you<br>
have all clients<br>
; behind a NAT, or for some other<br>
reason want Asterisk to<br>
; stay in the audio path, you may want<br>
to turn this off.<br>
<br>
; This setting also affect direct RTP<br>
; at call setup (a new feature in 1.4<br>
- setting up the<br>
; call directly between the endpoints<br>
instead of sending<br>
; a re-INVITE).<br>
<br>
; Additionally this option does not<br>
disable all reINVITE operations.<br>
; It only controls Asterisk generating<br>
reINVITEs for the specific<br>
; purpose of setting up a direct media<br>
path. If a reINVITE is<br>
; needed to switch a media stream to<br>
inactive (when placed on<br>
; hold) or to T.38, it will still be<br>
done, regardless of this<br>
; setting. Note that direct T.38 is<br>
not supported.<br>
<br>
<br>
<br>
<br>
Matt<br>
<span><font color="#888888"><br>
--<br>
Matthew Jordan<br>
Digium, Inc. | Engineering Manager<br>
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br>
Check us out at: <a href="http://digium.com" target="_blank">http://digium.com</a> & <a href="http://asterisk.org" target="_blank">http://asterisk.org</a><br>
<br>
--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
<a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
</font></span></blockquote></div><br><br clear="all"><br></div></div><div class="">-- <br><div>Regards</div>Sameer Rathod<div>8109413462 <br><div><br></div></div>
</div></div>
</blockquote></div><br><br clear="all"><br>-- <br><div>Regards</div>Sameer Rathod<div>8109413462 <br><div><br></div></div>
</div>