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