You need to understand that because FS is a B2BUA by default there are two separate media sessions endpoint1->fs and fs->endpoint2. They negotiate completely separately, and FS just copies media across during a bridge. That means endpoint2 never sees the SDP of endpoint1... unless you bypass_media. If you set bypass_media=true then they will see the actual SDP of the other endpoint (forwarded by FS). Reading RFC5245.it looks like that would be enough for it to work. But you'll still want to know *when* to bypass_media since it might not work in all situations especially when NAT is involved and when one endpoints doesn't support STUN/ICE. In a NAT situation you'll still be reliant on your router's SIP ALG being disabled - the endpoints are setting all their candidate addresses in the SDP and if the router were to modify them then that could break everything. In the case bypass_media_after_bridge=true might still allow RFC2833 DTMF for IVR menus hosted on FS before it leaves the media path. I'm not clear on how well ICE would work with the media renegotiation, but presumably it would work as well as the initial negotation. Chris, relying on SIP INFO isn't necessarily a good idea - not all endpoints would send it for a start, and it only ever made it as far as a IETF draft that was dropped... although plenty of people have implemented it it never became an actual standard. In particular there are timing issues with it if the packets arrive out of order due to network jitter, since it lacks the timestamping that RFC2833 provides. Far better to remain in the media path for IVR and then to bow out. You're going to be wanting to play prompts to the user during the IVR anyway. -Steve On 11 January 2013 12:47, Raimund Sacherer <raimund.sacherer@logitravel.com>wrote:
Hi,
I have not looked much into this, but I was under the impression that it's a client-side thing, as I read it the SDP contains various IP's for the media, the public (which the phone seeks out via STUN) and the private, the other phone tries to connect to each of those and the first answer it get's it uses for the media path, as a LAN should answer always faster than a WAN, it should create the mediapath directly between the two devices, therefore FS will not see the media and interacts only with the signaling.
This is how I have interpreted ICE.
Now, if it does not work this way I appologize, up to now I did not have any use for ICE as all our remote offices are small and they do normally not call each other and in our main offices our PBX's are (still) internal ...
Best, Rai
------------------------------ *From: *"Steven Ayre" <steveayre@gmail.com> *To: *"FreeSWITCH Users Help" <freeswitch-users@lists.freeswitch.org> *Sent: *Viernes, 11 de Enero 2013 13:34:15 *Subject: *Re: [Freeswitch-users] RTP traffic on LAn
Unfortunately FS does not (as yet) support it. At least not AFAIK. It also would require that both endpoints support it too, and not all do.
In any case since FS is a B2BUA not a SIP proxy the endpoints are negotiating with FS not the other endpoint - so you're still going to want to detect that you want to (try to) bypass_media.
-Steve
On 11 January 2013 11:19, Raimund Sacherer < raimund.sacherer@logitravel.com> wrote:
Hi,
isn't there allready a protocol that does that? ICE?
this is an excerpt from the SNOM wiki page (we have about 280 snom phones in action):
--- snip --- ICE DESCRIPTION Choose whether or not you want to use ICE (Interactive Connectivity Establishment). ICE optimizes the media path. This would be the case, for example, when two phones in the same network are calling each other via a long media path through other, external networks. With ICE, the short media path in the same network would be chosen, which will presumably have better quality than the long one. Sometimes this feature will stop you from being able to make calls. When this occurs, switch it off. --- snip ---
Link here: http://wiki.snom.com/Settings/user_ice
Here seems to be more information, check "Probing Media Path": http://wiki.snom.com/Networking/NAT
Best,
Raimund
------------------------------ *From: *"Chris B. Ware" <chrisbware@yahoo.it> *To: *freeswitch-users@lists.freeswitch.org *Sent: *Viernes, 11 de Enero 2013 10:30:52 *Subject: *[Freeswitch-users] RTP traffic on LAn
Hi all,
I'm trying to find a way to let RTP traffic between two phones, registered to a public Freeswitch, on the same LAN, remain local. Usually phones are natted behind an ADSL router and using two RTP streams to speack each other (because, for example, are on different rooms on the same office) consume a lot of bandwidth.
Is there anyone who has found a solution to this problem?
I was thinking on using SIP INFO for sending DTMF (in case customer have to use some phone features activated via phone), using private IP (no STUN on phones) and keeping FS out of RTP streaming.
Any help will be appreciated.
Chris
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