search for: pjmedia

...rc1 ./configure LDFLAGS="-z muldefs" --libdir=/usr/lib64 --with-unixodbc=$(odbc_config --include-prefix)/ --disable-dev-mode --with-pjproject-bundled ?checking for pjsip_dlg_create_uas_and_inc_lock in -lpjsip... no checking for pjsip_tsx_create_uac2 in -lpjsip... no checking if "pjmedia_mod_offer_flag flag = PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE" compiles using pjmedia.h... no checking for pjsip_get_dest_info in -lpjsip... no checking for pjsip/include/pjsip/sip_util.h in -lpj... no checking for pjsip_endpt_set_ext_resolver in -lpjsip... no checking if "struct pjsip_tls_set...

...so, with pjsip library. According to pjsip team's interpretation of g722, it works with 14bits PCM for input/output, so pjsip basically 'converts' the audio sample from 16 bits to 14 when encoding and vice-versa. Some implementations don't do 16<->14 bits conversion, so when pjmedia talks to one of those the over-driven audio problems appear. What we need to know is what's the most used implementation: 14<->16 bits conversion or not. Any pointers to help clear this up? We'd really like to see more g722-capable SIP clients for our own conference on ZipDX. Regar...

...INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: CSipSimple_d2vzw-16/r1916 Content-Type: application/sdp Content-Length: 354 v=0 o=- 3564161970 3564161970 IN IP4 172.17.9.1 s=pjmedia c=IN IP4 172.17.9.1 t=0 0 m=audio 4006 RTP/AVP 96 3 0 8 101 c=IN IP4 172.17.9.1 a=rtcp:4007 IN IP4 172.17.9.1 a=sendrecv a=rtpmap:96 SILK/8000 a=fmtp:96 useinbandfec=0 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------&...

..."ruvoip.net", nonce="1481885583/bcb53e85a740689479f116a96fc7086b", uri="sip:8xxx6yyy621 at txxx37.ru", response="843f8211896b5b05fcf3a633d6d8eedf", algori Content-Type: application/sdp Content-Length: 326 v=0 o=- 3690874445 3690874445 IN IP4 11.111.11.11 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 4022 RTP/AVP 0 8 109 96 c=IN IP4 11.111.11.11 b=TIAS:64000 a=rtcp:4023 IN IP4 11.111.11.11 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:109 G726-32/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 [2016-12-16 13:53:03] VERBOSE[13985] netsock2.c:...

...y Alexei Gradinari) * ASTERISK-28057 - chan_sip: SipNotify via AMI behaves differently to CLI (Reported by Peter Katzmann) * ASTERISK-28045 - configure script does not enforce libunbound2 version (Reported by Samuel Galarneau) * ASTERISK-28070 - testsuite: Sniffer assumes pjmedia will use ports below 10000 (Reported by Joshua C. Colp) * ASTERISK-27854 - rtp: Crash in off-nominal case where RTP instance can't be set up (Reported by Lei Fu) * ASTERISK-28059 - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by Joshua C. Colp)...

...ats that don't support seeking. ast_streamfile previously did unconditional seeking on files that broke playback of formats that don't support that functionality. This patch avoids the seek that was causing the problem. (closes issue ASTERISK-18994) Patched by: Timo Teras * Add pjmedia probation concepts to res_rtp_asterisk's learning mode. In order to better handle RTP sources with strictrtp enabled (which is the default setting in 10) using the learning mode to figure out new sources when they change is handled by checking for a number of consecutive (by sequence n...

...ERISK-26094 - stasis: Playing MOH to bridge with ARI does not work (Reported by Cameron) * ASTERISK-27920 - app_queue: Queue member considered inuse after immediately hanging up during dialing. (Reported by Cao Minh Hiep) * ASTERISK-28070 - testsuite: Sniffer assumes pjmedia will use ports below 10000 (Reported by Joshua C. Colp) * ASTERISK-28065 - res_odbc: missing SQL error diagnostic (Reported by Alexei Gradinari) * ASTERISK-27121 - res_pjsip_mwi: Memory leak on reload (Reported by Sergej Kasumovic) * ASTERISK-28059 - PJSIP: Up...

Hello you mean while placing a video call ? What info am I looking for in the debug output ? Kind regards. J. On 21-04-17 12:28, Marcelo Terres wrote: > Did you try to activate DEBUG and set the verbosity to a higher level > (100?) to check what Asterisk tells you about? > > Regards, > Marcelo H. Terres <mhterres at gmail.com> > IM: mhterres at

...y Alexei Gradinari) * ASTERISK-28057 - chan_sip: SipNotify via AMI behaves differently to CLI (Reported by Peter Katzmann) * ASTERISK-28045 - configure script does not enforce libunbound2 version (Reported by Samuel Galarneau) * ASTERISK-28070 - testsuite: Sniffer assumes pjmedia will use ports below 10000 (Reported by Joshua C. Colp) * ASTERISK-27854 - rtp: Crash in off-nominal case where RTP instance can't be set up (Reported by Lei Fu) * ASTERISK-28034 - chan_sip unstable with TLS after asterisk start or reloads (Reported by Davi...

...y Alexei Gradinari) * ASTERISK-28057 - chan_sip: SipNotify via AMI behaves differently to CLI (Reported by Peter Katzmann) * ASTERISK-28045 - configure script does not enforce libunbound2 version (Reported by Samuel Galarneau) * ASTERISK-28070 - testsuite: Sniffer assumes pjmedia will use ports below 10000 (Reported by Joshua C. Colp) * ASTERISK-27854 - rtp: Crash in off-nominal case where RTP instance can't be set up (Reported by Lei Fu) * ASTERISK-28034 - chan_sip unstable with TLS after asterisk start or reloads (Reported by Davi...

...y Alexei Gradinari) * ASTERISK-28057 - chan_sip: SipNotify via AMI behaves differently to CLI (Reported by Peter Katzmann) * ASTERISK-28045 - configure script does not enforce libunbound2 version (Reported by Samuel Galarneau) * ASTERISK-28070 - testsuite: Sniffer assumes pjmedia will use ports below 10000 (Reported by Joshua C. Colp) * ASTERISK-27854 - rtp: Crash in off-nominal case where RTP instance can't be set up (Reported by Lei Fu) * ASTERISK-28034 - chan_sip unstable with TLS after asterisk start or reloads (Reported by Davi...

...nn) - [ASTERISK-28045 <https://issues.asterisk.org/jira/browse/ASTERISK-28045>] - configure script does not enforce libunbound2 version (Reported by Samuel Galarneau) - [ASTERISK-28070 <https://issues.asterisk.org/jira/browse/ASTERISK-28070>] - testsuite: Sniffer assumes pjmedia will use ports below 10000 (Reported by Joshua C. Colp) - [ASTERISK-27854 <https://issues.asterisk.org/jira/browse/ASTERISK-27854>] - rtp: Crash in off-nominal case where RTP instance can't be set up (Reported by Lei Fu) - [ASTERISK-28034 <https://issues.asterisk.org/jir...