After struggling for 2 weeks, and maybe more than 50 hours spent over the internet.
Can someone tell me if he is able to have a very simple two softphones on the same network (no nat, same domain, no firewall) calling each other with the audio AND the video.
Phone1 is 192.168.1.15
Phone2 is 192.168.1.13
Asterisk server is 192.168.1.41 port 5060 UDP and TCP are activated and registration works with the two protocols.
I am using Raspbx (Asterisk 13.17.1, FreePBX 126.96.36.199) on a raspberry pi 3 out of the box. I tried Linphone, Zoiper.
I am under the feeling that the audio works amazingly well but the video is full of bugs… Am I trying to get something working that is not possible for the moment?
I do not understand why such a simple thing is so complicated… I tried with UDP, TCP, directmedia on and off, several codec…
Shall I try with asterisk 11?
If you are able to get the video working between two softphones could you post your files (sip_custom.conf and extension_custom.conf) and tell me which app and asterisk version do you use?
Thank you for your help, I am desperate…
vmexten=*97 useragent=FPBX-188.8.131.52(13.13.1) disallow=all allow=alaw allow=vp8 context=from-sip-external callerid=Unknown notifyringing=yes notifyhold=yes tos_sip=cs3 tos_audio=ef tos_video=af41 alwaysauthreject=yes limitonpeers=yes callevents=no rtpend=20000 tcpenable=yes context=from-sip-external rtpstart=10000 bindport=5060 jbenable=no rtpkeepalive=0 rtpholdtimeout=300 rtptimeout=30 srvlookup=no videosupport=yes tlsbindaddr=[::]:5061 tlsclientmethod=sslv2 tlsenable=no registertimeout=20 registerattempts=0 notifyringing=yes canreinvite=no checkmwi=10 defaultexpiry=120 g726nonstandard=no maxcallbitrate=384 maxexpiry=3600 minexpiry=60 notifyhold=yes allowguest=yes nat=force_rport,comedia ALLOW_SIP_ANON=no callerid=Unknown externip=77.XXX.XXX.107 localnet=192.168.1.0/24 language=en
[office-phone](!) ; create a template for our devices type=peer ; the channel driver will match on username first, IP second context=LocalSets ; this is where calls from the device will enter the dialplan host=dynamic ; the device will register with asterisk nat=no ; assume device is behind NAT ; *** NAT stands for Network Address Translation, which allows ; multiple internal devices to share an external IP address. secret=azerty ; a secure password for this device -- DON'T USE THIS PASSWORD! dtmfmode=auto ; accept touch-tones from the devices, negotiated automatically disallow=all ; reset which voice codecs this device will accept or offer ; which audio codecs to accept from, and request to, the device ; in the order we prefer transport=udp,tcp ; define a device name and use the office-phone template [Phone1](office-phone) allow=alaw allow=vp8 videosupport=yes ; define another device name using the same template [Phone2](office-phone) allow=alaw allow=vp8 videosupport=yes
[LocalSets] exten => 101,1,Dial(SIP/Phone1) exten => 102,1,Dial(SIP/Phone2) exten => 200,1,Answer() same => n,Wait(2) same => n,Playback(hello-world) same => n,Hangup()