Re: [asterisk-users] asterisk pjsip webrtc rtp to private IP

https://issues.asterisk.org/jira/browse/ASTERISK-28656 Dne 12/12/2019 v 16:49 Joshua C. Colp napsal(a): On Thu, Dec 12, 2019 at 11:40 AM marek > wrote: examples of "interesting" information like ICE result and howto make "minimal" configuration of

Re: [asterisk-users] asterisk pjsip webrtc rtp to private IP

On Thu, Dec 12, 2019 at 11:40 AM marek wrote: > examples of "interesting" information like ICE result and howto make > "minimal" configuration of pjproject.conf > > i.e. > > for debugging app_queue.so > > core set debug 5 app_queue.so > > for debugging RTP > > core set debug 10 rtp_engine >

Re: [asterisk-users] asterisk pjsip webrtc rtp to private IP

examples of "interesting" information like ICE result and howto make "minimal" configuration of pjproject.conf i.e. for  debugging app_queue.so core set debug 5 app_queue.so for debugging RTP core set debug 10 rtp_engine core set debug 10 res_rtp_asterisk rtp set debug on logger.conf rtp

Re: [asterisk-users] asterisk pjsip webrtc rtp to private IP

On Thu, Dec 12, 2019 at 10:57 AM marek wrote: > thank you very much. this is exactly whats needed for debug > > example output for your info > [Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: : > icess0x7f5d44081e88 .Added new remote candidate from the request: > 2.2.2.2:57536 > [Dec 12

Re: [asterisk-users] asterisk pjsip webrtc rtp to private IP

thank you very much. this is exactly whats needed for debug example output for your info [Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: : icess0x7f5d44081e88 .Added new remote candidate from the request: 2.2.2.2:57536 [Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: :  

Re: [asterisk-users] asterisk pjsip webrtc rtp to private IP

On Thu, Dec 12, 2019 at 8:57 AM marek wrote: > Asterisk is on public IP (as described in the first email) > > i have 10 years experience in voip, 4 years webrtc in production. i know > about ICE/STUN/DTLS-SRTP. yes, not every detail but the basic mechanism > > but i confess. i dont understand

Re: [asterisk-users] asterisk pjsip webrtc rtp to private IP

Asterisk is on public IP (as described in the first email) i have 10 years experience in voip, 4 years webrtc in production. i know about ICE/STUN/DTLS-SRTP. yes, not every detail but the basic mechanism but i confess. i dont understand WHY Asterisk SOMETIMES switches destination IP in RTP.

Re: [asterisk-users] asterisk pjsip webrtc rtp to private IP

On Thu, Dec 12, 2019 at 7:57 AM marek wrote: > with wireshark i need decrypt traffic every call which is time consuming. > get debug from pjnat through asterisk is not possible because of technical > reasons or nobody did it? > > > in my case its strange that ice candidates are the same > > good

Re: [asterisk-users] asterisk pjsip webrtc rtp to private IP

with wireshark i need decrypt traffic every call which is time consuming. get debug from pjnat through asterisk is not possible because of technical reasons or nobody did it? in my case its strange that ice candidates are the same good call v=0 o=- 3669976329745317845 2 IN IP4 127.0.0.1 s=-

Re: [asterisk-users] asterisk pjsip webrtc rtp to private IP

On Thu, Dec 12, 2019 at 6:39 AM marek wrote: > hi, > > i have following topology > > PSTN - Asterisk internet - router - jssip client (wss) > > Asterisk 13.29.1 on public IP, chan_pjsip for wss, chan_sip/udp for SIP > connection to PSTN > > router - public IP/private IP (NAT) > > jssip

[asterisk-users] asterisk pjsip webrtc rtp to private IP

hi, i have following topology PSTN - Asterisk internet -  router - jssip client (wss) Asterisk 13.29.1 on public IP, chan_pjsip for wss, chan_sip/udp for SIP connection to PSTN router - public IP/private IP (NAT) jssip client - private IP - sip over websocket to Asterisk PJSIP