[Date Prev][Date Next][Thread Prev][Thread Next][Date Index][Thread Index]
Re: [Linphone-developers] Linphone 3.8 STUN/ICE ignored on incomming cal
Re: [Linphone-developers] Linphone 3.8 STUN/ICE ignored on incomming call
Thu, 23 Apr 2015 00:06:47 +0200
Mozilla/5.0 (X11; Linux i686; rv:31.0) Gecko/20100101 Thunderbird/31.6.0
I am still looking into this problem as I could not achieve direct P2P
connections even with rtpengine on kamailio.
RTPengine should support ICE out of the box. And I confirm it supports,
but with other clients like Blink or CSipSimple which use their ICE
implementation. I could confirm this with seeing initial UDP traffic onn
server to be "released" and clients started to talk directly to each
other after a few seconds.
I could not reproduce this with Linphone and no RTP release does ever
occur. The clients continue to talk over the server, not p2p.
The one and only working setup is with using
in the kamailio config file. This actually does work for certain
scenarios, but it does not work reliably. RTP gets released always, but
not correctly and there is occasionaly no audio or video or both..
CSipSimple and Blink work with pure
without any flags set.
Can you please confirm, that the info in this thread
is still valid?
So ICE in Linphone works reliably ONLY with the linphone.org SIP
servers? What is different there from using kamailio + rtpengine?
Do I have to run my own TURN or STUN server too to get this working?
Shall there be any other module paramteres set (nathelper, rtpengine,
On 08.04.2015 17:46, jehan monnier wrote:
> Hi Filip,
> Both Asterix and Kamailio are supposed to be able to deal with SDP using
> private IP thanks to rtpproxy/relay and symmetric RTP.
> You don't need STUN to be enabled at Linphone level.
> By the way, you can try with our sip proxy
> at http://www.linphone.org/free-sip-service.html
> Best regards
> www.linphone.org <http://www.linphone.org>
> Le 8 avr. 2015 à 17:16, Filip Malenka <address@hidden
> <mailto:address@hidden>> a écrit :
>> Hi all..
>> meanwhile I switched from asterisk to kamailio sip proxy, because I
>> hoped for better NAT traversal mechanism, but even there it works only
>> with rtpproxy, nathelper modules and NAT enabled, so the media and RTP
>> packets get all effectively proxied over the server.
>> Is there any option to force Linphone (windows/linux/android) to use
>> STUN on incomming calls too?
>> As I wrote, at the moment it seems that STUN and public IP discovery
>> works only when doing outbound call.
>> This causes, that I can't reach any Linphones behind NAT directly and
>> SDP message contains wrong IP (192.168.1.xxx).
>> When I set the public IP manually in Linphone (windows/linux), calls
>> and direct RTP work as expected in both directions.
>> Thank you..
>> On 03.04.2015 23:07, Filip Malenka wrote:
>>> Hi all Linphone devs!
>>> Thank you all for your hard work contributing to the Linphone apps. They
>>> are awesome and I would like to contribute with pointing something out.
>>> What I want to achieve: To have Linphone clients on
>>> Windows/Ubuntu/Android all over, behind NATs, on mobile networks, etc.
>>> and to be able to reach every client from everywhere in every direction
>>> (Mobile -> NAT, NAT-> mobile, NAT-> another NAT, you can imagine..).
>>> I found out, that by setting my custom asterisk 11 to directrtpsetup=yes
>>> almost did the job. But there is a problem.
>>> When placing an outbound call from android -> NATed ubuntu, I get no RTP
>>> packets exchanged. The reason is, that Android on 3g network is trying
>>> to send to "192.168.1.xxx" my NATed network, which can't work. I was
>>> expecting that the Linphone from behind NAT would somehow propagate it's
>>> public IP through STUN/ICE to the android client first, but that's not
>>> If STUN option is selected in Linphone network settings, I would assume
>>> that every (inbound/outbound) call would be using it. Now only when
>>> doing outoubound calls, the STUN is utilized. Inbound call doesn't care
>>> about public IP. Thus the remote client sending RTP to wrong IP.
>>> I can get it work by manually setting my current public IP in Linphone.
>>> But this is not very "dynamic" workaround and I would need to set it
>>> everytime I change network/wifi/location or when public IP changes
>>> (occurs very often).. Not to say, that in android Linphone there is no
>>> such option to set IP manually.
>>> Is it possible to force doing a STUN request right before answering an
>>> inbound call?
>>> Maybe this isn't to be solved by Linphone client, but maybe an another
>>> asterisk setting. But I didn't figure it out yet.
>>> By setting asterisks
>>> nat=force_rport,comedia and
>>> nothing changed.
>> Linphone-developers mailing list
>> address@hidden <mailto:address@hidden>
> Linphone-developers mailing list