search for: pjnath

...put/pjlib-util-x86_64-unknown-linux-gnu/srv_resolver.o r - output/pjlib-util-x86_64-unknown-linux-gnu/string.o r - output/pjlib-util-x86_64-unknown-linux-gnu/stun_simple.o r - output/pjlib-util-x86_64-unknown-linux-gnu/stun_simple_client.o r - output/pjlib-util-x86_64-unknown-linux-gnu/xml.o ../src/pjnath/ice_strans.c: In function ?pj_ice_strans_sendto?: ../src/pjnath/ice_strans.c:1167:16: warning: variable ?pkt_size? set but not used [-Wunused-but-set-variable] ../src/pjnath/stun_msg.c: In function ?pj_stun_msg_decode?: ../src/pjnath/stun_msg.c:2303:14: warning: variable ?uattr_cnt? set but not use...

Hi , We trying to work on Asterisk 12 with sipml5+ PJNATH ,We can able register but calling is not working .The same configuration test with Xlite soft phone it working fine With Regards Prakash -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131211/9130c...

...ct bundled i have problem with segfault (centos 6) Program terminated with signal 11, Segmentation fault. #0 0xb7665695 in check_cached_response (sess=0xafbd688c, packet=0xb07676d8, pkt_size=132, options=1, token=0xafecc2bc, parsed_len=0x0, src_addr=0xb0e47a20, src_addr_len=16) at ../src/pjnath/stun_session.c:1287 1287 if (t->msg_magic == msg->hdr.magic && it was only once after 2 days. i dont know how to repeat it now :( any similiar experience? -- --------------------------------------- Marek Cervenka ======================================= -------------...

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. this is not only about ICE. its about RTP engine too which is Asterisk specific and Asterisk DEBUG is

Le lun. 26 août 2019 à 12:07, Joshua C. Colp <jcolp at digium.com> a écrit : > ... > > libpjnath is the ICE/STUN/TURN library which is used by res_rtp_asterisk > for that functionality. If you're using WebRTC or ICE/STUN/TURN, then you > would be using that library. > Yes, I'm using ICE/STUN/TURN. That explains libpjnath usage. Thank you sharing this here. Now finding why u...

...lly build Asterisk. I'm using the Asterisk pjproject 2.1 port from here: https://github.com/asterisk/pjproject An example of the build errors I'm resolving one by one is this: make[2]: *** No rule to make target `../../pjlib/lib/libpj-x86_64-unknown-linux-gnu.a', needed by `../lib/libpjnath-x86_64-unknown-linux-gnu.a'. Stop. make[1]: *** [/mnt/src/asterisk-11.4.0-rc1/res/pjproject/pjnath/lib/libpjnath-x86_64-unknown-linux-gnu.a] Error 2 make: *** [res] Error 2 I'm editing the Makefiles and fixing the paths so Asterisk can find the target. For all the people out there smarte...

...ic mechanism > > but i confess. i dont understand WHY Asterisk SOMETIMES switches > destination IP in RTP. this is not only about ICE. its about RTP > engine too which is Asterisk specific > > and Asterisk DEBUG is not helping > > > RTP traffic is given to pjnath to send using ICE, if this fails then > it uses the c= line. If you don't see (via ICE) then the fallback has > occurred and pjnath didn't send it via ICE, which most likely means > ICE negotiation failed for some reason. ICE and STUN is not encrypted > in Wireshark, so it c...

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 ~30% of calls has problem with no audio. reason is that Asterisk is sending RTP to private IP of jssip

...(Reported by Sebastian Kemper) * ASTERISK-28755 - SIP/Stasis: SIP headers not transmitted in the "variables" field (Reported by Jean Aunis - Prescom) * ASTERISK-28754 - ASTERISK-28738 Causes Audio Issue After Hold (Reported by Ross Beer) * ASTERISK-28716 - ICE: pjnath shouldn't wait for ICE to complete before allowing sending (Reported by Benjamin Keith Ford) * ASTERISK-28697 - res_pjsip: Named ACL does not update on reload if changed (Reported by Timothy Vanderaerden) * ASTERISK-28738 - Incorrect state machine used when...

...(Reported by Sebastian Kemper) * ASTERISK-28755 - SIP/Stasis: SIP headers not transmitted in the "variables" field (Reported by Jean Aunis - Prescom) * ASTERISK-28754 - ASTERISK-28738 Causes Audio Issue After Hold (Reported by Ross Beer) * ASTERISK-28716 - ICE: pjnath shouldn't wait for ICE to complete before allowing sending (Reported by Benjamin Keith Ford) * ASTERISK-28697 - res_pjsip: Named ACL does not update on reload if changed (Reported by Timothy Vanderaerden) * ASTERISK-28738 - Incorrect state machine used when...

...* ASTERISK-28697 - res_pjsip: Named ACL does not update on reload if changed (Reported by Timothy Vanderaerden) * ASTERISK-28746 - res_pjsip_outbound_registration keeps retrying the first entry in a SRV record set (Reported by George Joseph) * ASTERISK-28716 - ICE: pjnath shouldn't wait for ICE to complete before allowing sending (Reported by Benjamin Keith Ford) * ASTERISK-28738 - Incorrect state machine used when MOH_PASSTHRU is used (Reported by Torrey Searle) * ASTERISK-28742 - res_rtp_asterisk: static for audio due to i...

...* ASTERISK-28697 - res_pjsip: Named ACL does not update on reload if changed (Reported by Timothy Vanderaerden) * ASTERISK-28746 - res_pjsip_outbound_registration keeps retrying the first entry in a SRV record set (Reported by George Joseph) * ASTERISK-28716 - ICE: pjnath shouldn't wait for ICE to complete before allowing sending (Reported by Benjamin Keith Ford) * ASTERISK-28738 - Incorrect state machine used when MOH_PASSTHRU is used (Reported by Torrey Searle) * ASTERISK-28742 - res_rtp_asterisk: static for audio due to i...

...* ASTERISK-28697 - res_pjsip: Named ACL does not update on reload if changed (Reported by Timothy Vanderaerden) * ASTERISK-28746 - res_pjsip_outbound_registration keeps retrying the first entry in a SRV record set (Reported by George Joseph) * ASTERISK-28716 - ICE: pjnath shouldn't wait for ICE to complete before allowing sending (Reported by Benjamin Keith Ford) * ASTERISK-28738 - Incorrect state machine used when MOH_PASSTHRU is used (Reported by Torrey Searle) * ASTERISK-28742 - res_rtp_asterisk: static for audio due to i...

...* ASTERISK-28697 - res_pjsip: Named ACL does not update on reload if changed (Reported by Timothy Vanderaerden) * ASTERISK-28746 - res_pjsip_outbound_registration keeps retrying the first entry in a SRV record set (Reported by George Joseph) * ASTERISK-28716 - ICE: pjnath shouldn't wait for ICE to complete before allowing sending (Reported by Benjamin Keith Ford) * ASTERISK-28738 - Incorrect state machine used when MOH_PASSTHRU is used (Reported by Torrey Searle) * ASTERISK-28742 - res_rtp_asterisk: static for audio due to i...

...K-26692 - res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip) (Reported by scgm11) * ASTERISK-26835 - res_rtp_asterisk: Crash when freeing RTCP address string (Reported by Niklas Larsson) * ASTERISK-26853 - res_rtp_asterisk: Crash in pjnath when receiving packet (Reported by Adagio) * ASTERISK-26613 - format_wav: wav16 format read file only by 320 - half of frame (Reported by Vitaly K) * ASTERISK-26169 - format_ogg_vorbis: Memory leak using OGG in MixMonitor (Reported by Ivan Myalkin) * ASTERISK-...

...K-26692 - res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip) (Reported by scgm11) * ASTERISK-26835 - res_rtp_asterisk: Crash when freeing RTCP address string (Reported by Niklas Larsson) * ASTERISK-26853 - res_rtp_asterisk: Crash in pjnath when receiving packet (Reported by Adagio) * ASTERISK-26613 - format_wav: wav16 format read file only by 320 - half of frame (Reported by Vitaly K) * ASTERISK-26169 - format_ogg_vorbis: Memory leak using OGG in MixMonitor (Reported by Ivan Myalkin) * ASTERISK-...

I have a CentOS 6.3 machine I installed Asterisk 11, worked fine... I then tried to install on Cents 5.8, seemed to go fine... Then when I placed a call I got this: ast_rtp_instance_new: No RTP engine was found. Do you have one loaded? Did a search and found issues with ARM and this problem but did not help me, not using gtalk or anything. Just call between two polycom phones on local network.

..._asterisk.c but i'm not sure where to put > DEBUG info about which IP and why Asterisk pick for RTP > > any hint? > > > is it possible debug Asterisk STUN request/response ? or is it > hidden in > pjsip internals? > > > ICE is performed using pjnath, which is part of pjproject and not > Asterisk itself. Looking at the SDP and the ICE candidates can tell > you the possibilities for the paths, and a wireshark capture can show > you the actual traffic going back and forth and what is being attempted. > > -- > Joshua C. Colp...

...* ASTERISK-28697 - res_pjsip: Named ACL does not update on reload if changed (Reported by Timothy Vanderaerden) * ASTERISK-28746 - res_pjsip_outbound_registration keeps retrying the first entry in a SRV record set (Reported by George Joseph) * ASTERISK-28716 - ICE: pjnath shouldn't wait for ICE to complete before allowing sending (Reported by Benjamin Keith Ford) * ASTERISK-28738 - Incorrect state machine used when MOH_PASSTHRU is used (Reported by Torrey Searle) * ASTERISK-28742 - res_rtp_asterisk: static for audio due to i...

..._rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip) (Reported by Sebastian Gutierrez) * ASTERISK-26835 - res_rtp_asterisk: Crash when freeing RTCP address string (Reported by Niklas Larsson) * ASTERISK-26853 - res_rtp_asterisk: Crash in pjnath when receiving packet (Reported by Adagio) * ASTERISK-26613 - format_wav: wav16 format read file only by 320 - half of frame (Reported by Vitaly K) * ASTERISK-26169 - format_ogg_vorbis: Memory leak using OGG in MixMonitor (Reported by Ivan Myalkin) * ASTERISK-...