PJ SIP
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- Re: How to know incoming) call is audio or video
- From: Mu-Sheng Wu <mu-sheng@xxxxxxxxxxxx>
- SDP offer contains to audio media descriptions, only the first one is active
- From: Bogdan Cristea <cristeab@xxxxxxxxx>
- Re: pjsip put android phone on speaker
- From: Mu-Sheng Wu <mu-sheng@xxxxxxxxxxxx>
- Outbound hang up problem
- From: Musheng Wu <mu-sheng@xxxxxxxxxxxx>
- I need a video call tutorial on Android.
- From: Musheng Wu <mu-sheng@xxxxxxxxxxxx>
- Roadmap - release-2.7
- From: Gabor K <gabor58k@xxxxxxxxx>
- How to get remote pick up event?
- From: Gary <magiclin99@xxxxxxxxx>
- Re: Memory leak with OPTIONS dialogs
- From: mscdexdotexe <mscdex@xxxxxxxxxx>
- Re: dyld: Library not loaded: ios_local/lib/libopenh264.4.dylib
- From: Wasim Mirza <wasim.mirza@xxxxxxxxxxxx>
- dyld: Library not loaded: ios_local/lib/libopenh264.4.dylib
- From: Wasim Mirza <wasim.mirza@xxxxxxxxxxxx>
- Re: Memory leak with OPTIONS dialogs
- From: mscdexdotexe <mscdex@xxxxxxxxxx>
- Re: Ynt: How to get remote hold event on local ?
- Re: Ynt: How to get remote hold event on local ?
- From: "Tailor, Mayur" <mayur.tailor@xxxxxxxxxx>
- Ynt: How to get remote hold event on local ?
- From: qstn q <qstn@xxxxxxxxxxx>
- Receive audio data from 2 different streams
- From: Bogdan Cristea <cristeab@xxxxxxxxx>
- Re: Memory leak with OPTIONS dialogs
- From: mscdexdotexe <mscdex@xxxxxxxxxx>
- Removing UPDATE tag from Allow header
- From: qstn q <qstn@xxxxxxxxxxx>
- Memory leak with OPTIONS dialogs
- From: mscdexdotexe <mscdex@xxxxxxxxxx>
- How to get remote hold event on local ?
- From: "Tailor, Mayur" <mayur.tailor@xxxxxxxxxx>
- pjsua2 how to parse one particular header and get its value when incoming call?
- From: "zyxsjl@xxxxxxx" <zyxsjl@xxxxxxx>
- Show green layer on android surface view
- From: "Tailor, Mayur" <mayur.tailor@xxxxxxxxxx>
- pjsip video call :hangup videocall will block thread when network disconnected
- From: "weili.wu" <wuweili@xxxxxxxxxxx>
- Re: PJSIP Android put on Speaker
- From: "Tailor, Mayur" <mayur.tailor@xxxxxxxxxx>
- Re: PJSIP Android put on Speaker
- From: Andrzej Grajnert <frogersik@xxxxxxxxx>
- PJSIP Android put on Speaker
- From: "Tailor, Mayur" <mayur.tailor@xxxxxxxxxx>
- Undefined reference error when building pjsip with TARGET_ABI=armeabi-v7a
- From: Gary <magiclin99@xxxxxxxxx>
- Re: [PATCH] Fix Speex Resampler multi channel mode
- From: Tobias Schlager <Tobias.Schlager@xxxxxxxxxxxxx>
- Re: [PATCH] Fix Speex Resampler multi channel mode
- From: Nanang Izzuddin <nanang@xxxxxxxxx>
- Re: [PATCH] Add GnuTLS, DTLS, and other improvements
- From: Alexandre Viau <alexandre.viau@xxxxxxxxxxxxxxxxxxxx>
- pjsua python with no audio
- From: Anders Finn <anders@xxxxxxxxxxxxxxxx>
- Conference connection level implemented (with patch)
- From: Michael Scheiffler <michael@xxxxxxxxxxxxx>
- [PATCH] Fix MMDevice cross compilation
- From: Tobias Schlager <Tobias.Schlager@xxxxxxxxxxxxx>
- [PATCH] Allow configuration of the Speex resampler instead of libsamplerate
- From: Tobias Schlager <Tobias.Schlager@xxxxxxxxxxxxx>
- Re: [PATCH] Replace Windows-specific types in Speex resampler
- From: Tobias Schlager <Tobias.Schlager@xxxxxxxxxxxxx>
- [PATCH] Fix Speex Resampler multi channel mode
- From: Tobias Schlager <Tobias.Schlager@xxxxxxxxxxxxx>
- [PATCH] Replace Windows-specific types in Speex resampler
- From: Tobias Schlager <Tobias.Schlager@xxxxxxxxxxxxx>
- Re: Error when processing reply from server with userless account
- From: Andreas Wehrmann <a.wehrmann@xxxxxxxxxx>
- Error when processing reply from server with userless account
- From: Bogdan Cristea <cristeab@xxxxxxxxx>
- PJSUA2 Sip Server
- From: Salih YILDIRIM <salih.yildirim@xxxxxxxxxx>
- pjsip put android phone on speaker
- From: "Tailor, Mayur" <mayur.tailor@xxxxxxxxxx>
- support for precondition
- From: frogersik <frogersik@xxxxxxxxx>
- Re: Bad RTP pt 104 (expecting 9) + random source warning
- From: Andreas Wehrmann <a.wehrmann@xxxxxxxxxx>
- Opus Audio codec with pjsip not working on android
- From: "Tailor, Mayur" <mayur.tailor@xxxxxxxxxx>
- Re: Bad RTP pt 104 (expecting 9) + random source warning
- From: Kevin Rombach via pjsip <pjsip@xxxxxxxxxxxxxxx>
- Re: Bad RTP pt 104 (expecting 9) + random source warning
- From: Andreas Wehrmann <a.wehrmann@xxxxxxxxxx>
- Re: Bad RTP pt 104 (expecting 9) + random source warning
- From: Kevin Rombach via pjsip <pjsip@xxxxxxxxxxxxxxx>
- Re: Bad RTP pt 104 (expecting 9) + random source warning
- From: Andreas Wehrmann <a.wehrmann@xxxxxxxxxx>
- Re: Bad RTP pt 104 (expecting 9) + random source warning
- From: Kevin Rombach via pjsip <pjsip@xxxxxxxxxxxxxxx>
- Re: Bad RTP pt 104 (expecting 9) + random source warning
- From: Andreas Wehrmann <a.wehrmann@xxxxxxxxxx>
- Re: Bad RTP pt 104 (expecting 9) + random source warning
- From: Kevin Rombach via pjsip <pjsip@xxxxxxxxxxxxxxx>
- Re: Bad RTP pt 104 (expecting 9) + random source warning
- From: Andreas Wehrmann <a.wehrmann@xxxxxxxxxx>
- Splitcomb and audio delay
- Why is PJSIP 2.5.5 not sending ACK after Asterisk sends the 200 OK packet
- From: Aleksandar Milenkovic <shark@xxxxxxxxxxxxxx>
- Re: Bad RTP pt 104 (expecting 9) + random source warning
- From: Kevin Rombach via pjsip <pjsip@xxxxxxxxxxxxxxx>
- Unable to find default audio device
- From: Kevin Rombach via pjsip <pjsip@xxxxxxxxxxxxxxx>
- Re: Bad RTP pt 104 (expecting 9) + random source warning
- From: Andreas Wehrmann <a.wehrmann@xxxxxxxxxx>
- Bad RTP pt 104 (expecting 9) + random source warning
- From: Kevin Rombach via pjsip <pjsip@xxxxxxxxxxxxxxx>
- how to set WMME audio device
- From: segalion <segalion@xxxxxxxxx>
- Re: RTP-Rx statistic not plausible if decoder is paused
- From: Michael Scheiffler <michael@xxxxxxxxxxxxx>
- R: Set device Volume with Pjmedia
- From: "Lele" <86eldnl@xxxxxxxxx>
- ice_demo doesn't use TURN/STUN server? How can it do NAT Traversal?
- From: Chip Burwell <chipburwell@xxxxxxxxx>
- Re: Crash with TURN enabled caused by an apparent race condition
- From: Alex Solis <alex.solis@xxxxxxxxxxxxxx>
- Set device Volume with Pjmedia
- From: "Lele" <86eldnl@xxxxxxxxx>
- Re: RTP Extension Header
- From: Лухнов Андрей Олегович <loukhnov@xxxxxxxxxxx>
- RTP Extension Header
- From: Simone Freddio <simone.freddio@xxxxxxxxxxxxxxx>
- SDP getting configured with site IP instead of global IP in IPV6.
- From: "vivek@xxxxxxxxxxxxxxxx" <vivek@xxxxxxxxxxxxxxxx>
- Re: Assertion in sip_endpoint.c:170: pjsip_endpt_register_module
- From: "Marant, Jerome (External)" <jerome.marant.external@xxxxxxxxxx>
- Assertion in sip_endpoint.c:170: pjsip_endpt_register_module
- From: "Marant, Jerome (External)" <jerome.marant.external@xxxxxxxxxx>
- "481 Call/Transaction Does Not Exist" when canceling before call connects
- From: Danyi Lin <Danyi@xxxxxxxxxxxx>
- undefined reference to `pjsua_create'
- From: Viditha Murahari <viditha@xxxxxxxxxxxxxxxxx>
- PjSua2 with .Net Threading Problem
- From: Salih YILDIRIM <salih.yildirim@xxxxxxxxxx>
- Re: Create more than one pj_sock_socket for multicast connection - How to manage memory pool of pjmedia_endpoint
- From: Alain Totouom <alain.totouom@xxxxxx>
- R: Create more than one pj_sock_socket for multicast connection - How to manage memory pool of pjmedia_endpoint
- From: "Lele" <86eldnl@xxxxxxxxx>
- Re: Create more than one pj_sock_socket for multicast connection - How to manage memory pool of pjmedia_endpoint
- From: Alain Totouom <alain.totouom@xxxxxx>
- Create more than one pj_sock_socket for multicast connection - How to manage memory pool of pjmedia_endpoint
- From: "Lele" <86eldnl@xxxxxxxxx>
- Unit tests in android
- From: Symeon Mattes <symeon.mattes@xxxxxxxxx>
- Re: Create more than one pj_sock_socket for multicast connection - How to manage memory pool of pjmedia_endpoint
- From: Alain Totouom <alain.totouom@xxxxxx>
- Re-registration problem with expires less than PJSIP_REGISTER_CLIENT_DELAY_BEFORE_REFRESH
- From: Alexey Ermoshin <alexey.ermoshin.78@xxxxxxxxx>
- Application-Specific RTCP packets
- From: "Lele" <86eldnl@xxxxxxxxx>
- Bug in account handling
- From: Martin Schmid <scm@xxxxxxxxxxxxxx>
- Re: Video Call In Background
- From: Oren Barash <oren.elbitsystems@xxxxxxxxx>
- RTP-Rx statistic not plausible if decoder is paused
- From: Michael Scheiffler <michael@xxxxxxxxxxxxx>
- Re: Video Call In Background
- From: Vedvyas Rauniyar <vedvyas.rauniyar@xxxxxxxxxxx>
- Re: Video Call In Background
- From: Oren Barash <oren.elbitsystems@xxxxxxxxx>
- Re: Video Call In Background
- From: Vedvyas Rauniyar <vedvyas.rauniyar@xxxxxxxxxxx>
- Re: Memory Leak problem with Pool
- From: 刘猛 <meng.liu@xxxxxxxxxxxx>
- Create more than one pj_sock_socket for multicast connection - How to manage memory pool of pjmedia_endpoint
- From: "Lele" <86eldnl@xxxxxxxxx>
- Conference slot issue with simultaneous call end / call begin
- Video Call In Background
- From: Vedvyas Rauniyar <vedvyas.rauniyar@xxxxxxxxxxx>
- Re: Memory Leak problem with Pool
- From: Yuming Zheng <zhengyumingnanjing@xxxxxxxxx>
- Connect on multicast ipaddress (send/receive rtp packets)
- From: "Lele" <86eldnl@xxxxxxxxx>
- Re: Memory Leak problem with Pool
- From: 刘猛 <meng.liu@xxxxxxxxxxxx>
- Set PJMEDIA codec in PJMEDIA ENDPOINT
- From: "Lele" <86eldnl@xxxxxxxxx>
- Re: Pjsip Client registration
- From: Nahum Nir <hello.shalom.hi@xxxxxxxxx>
- Pjsip Client registration
- From: alaa <alaa@xxxxxxxxxxx>
- Connection on a Multicast channel with PJSIP
- From: "Lele" <86eldnl@xxxxxxxxx>
- TLS - Changing video stream direction corrupt the video stream
- From: Oren Barash <oren.elbitsystems@xxxxxxxxx>
- Re: Memory Leak problem with Pool
- From: Yuming Zheng <zhengyumingnanjing@xxxxxxxxx>
- Re: how to disable buffering in pjsua log file in linux?
- From: Bill Gardner <billg@xxxxxxxxxxxx>
- how to disable buffering in pjsua log file in linux?
- From: segalion <segalion@xxxxxxxxx>
- Memory Leak problem with Pool
- From: 刘猛 <meng.liu@xxxxxxxxxxxx>
- Re: Assigning a value to a pj_str_t
- From: "Marant, Jerome (External)" <jerome.marant.external@xxxxxxxxxx>
- Re: Assigning a value to a pj_str_t
- From: Bill Gardner <billg@xxxxxxxxxxxx>
- Does PJSIP mute early dialog when the UAC receive several SIP 183 for early media?
- From: Rodrigo Pimenta Carvalho <pimenta@xxxxxxxxx>
- Assigning a value to a pj_str_t
- From: "Marant, Jerome (External)" <jerome.marant.external@xxxxxxxxxx>
- Re: Is it possible to fork early media?
- From: Rodrigo Pimenta Carvalho <pimenta@xxxxxxxxx>
- Re: SIP INVITE/ACK and SDP (2)
- From: "Marant, Jerome (External)" <jerome.marant.external@xxxxxxxxxx>
- Is it possible to fork early media?
- From: Rodrigo Pimenta Carvalho <pimenta@xxxxxxxxx>
- Re: SIP INVITE/ACK and SDP (2)
- From: Alain Totouom <alain.totouom@xxxxxx>
- Re: SIP INVITE/ACK and SDP (2)
- From: "Marant, Jerome (External)" <jerome.marant.external@xxxxxxxxxx>
- Re: SIP INVITE/ACK and SDP (2)
- From: "Marant, Jerome (External)" <jerome.marant.external@xxxxxxxxxx>
- Re: SIP INVITE/ACK and SDP (2)
- From: Alain Totouom <alain.totouom@xxxxxx>
- Doxygen LaTeX documentation troubles fixed
- From: Michael Scheiffler <michael@xxxxxxxxxxxxx>
- SIP INVITE/ACK and SDP (2)
- From: "Marant, Jerome (External)" <jerome.marant.external@xxxxxxxxxx>
- SIP INVITE/ACK and SDP
- From: "Marant, Jerome (External)" <jerome.marant.external@xxxxxxxxxx>
- Re: undefined reference to `pj_dns_parse_addr_response'
- From: KY <gv.kylin@xxxxxxxxx>
- More than 2 simultaneous audio devices
- pjsip python - compilation problem
- From: "Bc. Josef Jebavy" <josef.jebavy@xxxxxxxx>
- how to save pjsip video as picture
- From: "jeff" <jeff.yc.chen@xxxxxxxxxxxx>
- Android Sample Problem: Incoming & Outgoing Calls
- From: Tunç Ikikardes <tunch.ikikardes@xxxxxxxxx>
- ICE is changing IP in SDP , from private to public. It seems wrong.
- From: Rodrigo Pimenta Carvalho <pimenta@xxxxxxxxx>
- How to process arbitrary request
- From: Sergei Meishutovich <Sergei.Meishutovich@xxxxxxxxxxx>
- Re: [PATCH] Add GnuTLS, DTLS, and other improvements
- From: Adonay Felipe Nogueira <adfeno@xxxxxxxxxxxxxxx>
- assertion failed error when answer in coming call with pjsip_sc_busy_here(486)
- From: "zyxsjl@xxxxxxx" <zyxsjl@xxxxxxx>
- Echo cancellation for windows platform
- From: "R, Rajkumar (Raj)" <rajkumaradass@xxxxxxxxx>
- Handling ip change during call
- From: John Gathm <john.gathm@xxxxxxxxx>
- Re: VAD in pjmedia
- From: Gabor K <gabor58k@xxxxxxxxx>
- Re: How to know incoming) call is audio or video
- From: Oren Barash <oren.elbitsystems@xxxxxxxxx>
- How to vad...
- From: Gabor K <gabor58k@xxxxxxxxx>
- Symmetric NAT and port forwarding
- From: Thiago Guimarães <thiago.barcelos@xxxxxxxxx>
- Re: Audio Underflow's effect on latency
- From: Bill Gardner <billg@xxxxxxxxxxxx>
- Re: Audio Underflow's effect on latency
- From: John M <jdm0x0430@xxxxxxxxx>
- Re: Audio Underflow's effect on latency
- From: Bill Gardner <billg@xxxxxxxxxxxx>
- Audio Underflow's effect on latency
- From: John M <jdm0x0430@xxxxxxxxx>
- Objective-C compiler not installed on this system
- From: Nizar Ellouze <nizarellouze@xxxxxxxx>
- Re: Video Keep Alive is Disabled
- From: Oren Barash <oren.elbitsystems@xxxxxxxxx>
- Re: Missing initializer in PJMedia mips_test.c
- From: Ming <ming@xxxxxxxxx>
- Re: Objective-C compiler needed for Android Build
- From: Ming <ming@xxxxxxxxx>
- Re: Missing pragma pack causes gcov crash
- From: Ming <ming@xxxxxxxxx>
- Re: Objective-C compiler needed for Android Build
- From: Tunç Ikikardes <tunch.ikikardes@xxxxxxxxx>
- Objective-C compiler needed for Android Build
- From: Tunç Ikikardes <tunch.ikikardes@xxxxxxxxx>
- Missing initializer in PJMedia mips_test.c
- From: Michael Scheiffler <michael@xxxxxxxxxxxxx>
- Missing pragma pack causes gcov crash
- From: Michael Scheiffler <michael@xxxxxxxxxxxxx>
- How to know incoming) call is audio or video
- From: "Tailor, Mayur" <mayur.tailor@xxxxxxxxxx>
- Immediate Hiring -freeswitch Developers- Noida/Delhi-India
- From: sales <sales@xxxxxxxxxxxxxxxxx>
- Detecting SRTP of Remote
- From: Peter Warrick <peter@xxxxxxxxxxxxxx>
- Unable to stablish call through SIPS with SRTP Enforced.
- From: Gabriel Vasconcelos <gfreivasc@xxxxxxxxx>
- pjsua 2.6 does not unsubscribe mwi at quit
- From: Juha Heinanen <jh@xxxxxxxxxx>
- Contact header when SRTP is optional
- From: David Talmage <sip.phone.fan@xxxxxxxxx>
- Python video call
- From: Boštjan Komac <bostjan68@xxxxxxxxx>
- Change video codec params during a video call
- From: Mark Shen <subscribeit@xxxxxxxxx>
- Re: [PATCH] Add GnuTLS, DTLS, and other improvements
- From: Nanang Izzuddin <nanang@xxxxxxxxx>
- Re: Patch for crash in pjsua2 pj2Str()
- From: Nanang Izzuddin <nanang@xxxxxxxxx>
- IP in SDP from PJSIP UA changes only after router reset. Why?
- From: Rodrigo Pimenta Carvalho <pimenta@xxxxxxxxx>
- How to add body to SIP REGISTRATION message in PJSIP using JAVA or C++?
- Re: How to put Proxy-Authorization in first SIP INVITE?
- From: Rodrigo Pimenta Carvalho <pimenta@xxxxxxxxx>
- Re: How to put Proxy-Authorization in first SIP INVITE?
- From: Andrzej Grajnert <frogersik@xxxxxxxxx>
- Video Keep Alive is Disabled
- From: Oren Barash <oren.elbitsystems@xxxxxxxxx>
- Re: How to put Proxy-Authorization in first SIP INVITE?
- From: Gang Liu <gangban.lau@xxxxxxxxx>
- How to put Proxy-Authorization in first SIP INVITE?
- From: Rodrigo Pimenta Carvalho <pimenta@xxxxxxxxx>
- memory player and memory recorder
- From: Cucumber <yacucumber@xxxxxxxxx>
- [PATCH] Add GnuTLS, DTLS, and other improvements
- From: Adonay Felipe Nogueira <adfeno@xxxxxxxxxxxxxxx>
- Patch for crash in pjsua2 pj2Str()
- From: Nick Dowell <nick@xxxxxxxxxxxxxx>
- Debug PJSIP with android studio
- From: "R, Rajkumar (Raj)" <rajkumaradass@xxxxxxxxx>
- Re: Building pjsip 2.6 for armeabi-v7a with Android NDK
- From: "Tailor, Mayur" <mayur.tailor@xxxxxxxxxx>
- Re: Video calls on Android - which libs are required?
- From: Žarko Coklin <zcoklin@xxxxxxxxxxx>
- Speex AEC reducing audio it shouldn't
- From: John M <jdm0x0430@xxxxxxxxx>
- Re: Building pjsip 2.6 for armeabi-v7a with Android NDK
- From: Nick Dowell <nick@xxxxxxxxxxxxxx>
- Building pjsip 2.6 for armeabi-v7a with Android NDK
- From: "Tailor, Mayur" <mayur.tailor@xxxxxxxxxx>
- Problem compiling PJSIP: recompile with -fPIC
- From: Oded Arbel <oded.arbel@xxxxxxxxxxxxxxxxxx>
- PJSIP - iOS external sound device
- From: Owen L Brown <owen@xxxxxxxxxxx>
- STUN in local network
- From: Thiago Guimarães <thiago.barcelos@xxxxxxxxx>
- pjsip INVITE with URN support
- From: Andrzej Grajnert <frogersik@xxxxxxxxx>
- Re: asterisk advisories AST-2017-002 and AST-2017-003
- From: Ming <ming@xxxxxxxxx>
- asterisk advisories AST-2017-002 and AST-2017-003
- From: Tzafrir Cohen <tzafrir.cohen@xxxxxxxxxx>
- Re: STUN in local network
- From: Nahum Nir <hello.shalom.hi@xxxxxxxxx>
- Resolve attempt to JIT for PJSIP (PJSUA2) C# on Xamarin.iOS AOT only
- From: Casey Lowe <casey.lowe.viasat@xxxxxxxxx>
- Re: RTP Timeout Detection
- From: Piewald Georg <gpi@xxxxxxxxxxxxxxxxxxxx>
- Re: RTP Timeout Detection
- From: JOHAN LANTZ <johan.lantz@xxxxxxxxxxxxxx>
- RTP Timeout Detection
- From: Piewald Georg <gpi@xxxxxxxxxxxxxxxxxxxx>
- Eraly session support in pjsip_inv_session (2 SDPs in one SIP message)
- From: Zoltan.Toth.ext@xxxxxxxxxxxxxxxxx
- STUN in local network
- From: Thiago Guimarães <thiago.barcelos@xxxxxxxxx>
- Antwort: [Newsletter] parse error with IPV6 addresses in square brackets
- From: Zoltan.Toth.ext@xxxxxxxxxxxxxxxxx
- parse error with IPV6 addresses in square brackets
- From: Franz Georg Köhler <lists@xxxxxxxxxxx>
- Poor audio quality with splitter combiner
- From: John M <jdm0x0430@xxxxxxxxx>
- Re: pjsip_inv_send_msg segfault fix
- From: Nanang Izzuddin <nanang@xxxxxxxxx>
- Re: unregister_and_destroy_dialog: unset dlg->dlg_set
- From: Nanang Izzuddin <nanang@xxxxxxxxx>
- Integrating PJSIP 2.6 (PJSUA2) in Xamarin.iOS (C#) [TypeInitializationException]
- From: Casey Lowe <casey.lowe.viasat@xxxxxxxxx>
- Re: NAT64 ios issue
- From: Girish Gopinath <gopinath_girish@xxxxxxxxxxx>
- Re: NAT64 ios issue
- From: JOHAN LANTZ <johan.lantz@xxxxxxxxxxxxxx>
- Re: NAT64 ios issue
- From: Girish Gopinath <gopinath_girish@xxxxxxxxxxx>
- Re: compile error on mips platform
- From: Seanchann Zhou <zhouxiaoqiang.mstech@xxxxxxxxx>
- compile error on mips platform
- From: Seanchann Zhou <zhouxiaoqiang.mstech@xxxxxxxxx>
- Re: pjsip_inv_send_msg segfault fix
- From: Alexei Gradinari <alex2grad@xxxxxxxxx>
- unregister_and_destroy_dialog: unset dlg->dlg_set
- From: Alexei Gradinari <alex2grad@xxxxxxxxx>
- pjsip_inv_send_msg segfault fix
- From: Alexei Gradinari <alex2grad@xxxxxxxxx>
- Problem with Toshiba gateway
- From: Michael Goffioul <michael.goffioul@xxxxxxxxx>
- STUN in local network
- From: Thiago Guimarães <thiago.barcelos@xxxxxxxxx>
- Android phone audio problem
- From: Branko Zebec <branko.zebec@xxxxxxxxx>
- Re: NAT64 ios issue
- From: Girish Gopinath <gopinath_girish@xxxxxxxxxxx>
- Re: NAT64 ios issue
- From: JOHAN LANTZ <johan.lantz@xxxxxxxxxxxxxx>
- Re: NAT64 ios issue
- From: Girish Gopinath <gopinath_girish@xxxxxxxxxxx>
- Android phone audio problem
- From: Branko Zebec <branko.zebec@xxxxxxxxx>
- Android device how to change between handset and speakerbox
- From: Branko Zebec <branko.zebec@xxxxxxxxx>
- Enabling compact headers per call
- From: David Talmage <sip.phone.fan@xxxxxxxxx>
- Pulse Audio in Pjsip
- From: ABHILASH M J <abhilash_j@xxxxxxxxx>
- clock_thread() sleep deeply after illegal/malformed RTP packages received
- From: Xiaoming Deng <dengxiaoming@xxxxxxxxx>
- Android Issue on Ticket#1861(Video capture orientation)
- From: Xiaoming Deng <dengxiaoming@xxxxxxxxx>
- Callback function to calculate the response digest in java library
- Increase timeout for processing outgoing call
- From: bbnnvv <bbnnvv@xxxxx>
- Re: Pulse Audio in Pjsip
- From: ABHILASH M J <abhilash_j@xxxxxxxxx>
- Re: Pulse Audio in Pjsip
- From: sales <sales@xxxxxxxxxxxxxxxxx>
- Pulse Audio in Pjsip
- From: ABHILASH M J <abhilash_j@xxxxxxxxx>
- PjSip transport
- From: Denis Zhuchinski <denys.zhuchinsky@xxxxxxxxx>
- Re: android random crash when client call heavily
- From: W00HA09.高宇(万睿) <gaoyu01@xxxxxxxxx>
- PJSUA2 in Java: Multiple audio device
- From: Simone Freddio <simone.freddio@xxxxxxxxxxxxxxx>
- PJSUA2 in Java: Multiple audio device
- From: Simone Freddio <simone.freddio@xxxxxxxxxxxxxxx>
- Change caller ID before making a call
- From: Bogdan Cristea <cristeab@xxxxxxxxx>
- symbol lookup error with Python library
- From: Oded Arbel <oded@xxxxxxxxxx>
- Re: android random crash when client call heavily
- From: frogersik <frogersik@xxxxxxxxx>
- Re: Bug in SRTP negotiation
- From: Riza Sulistyo <riza@xxxxxxxxx>
- Re: base64: fix issues
- From: Ming <ming@xxxxxxxxx>
- PJSIP Python sound device name
- From: Антон Каменский <antkamidiv@xxxxxxx>
- base64: fix issues
- From: Adrien Béraud <adrien.beraud@xxxxxxxxxxxxxxxxxxxx>
- Re: Assertion failed: (a->addr.sa_family == PJ_AF_INET || a->addr.sa_family == PJ_AF_INET6)
- From: Piewald Georg <gpi@xxxxxxxxxxxxxxxxxxxx>
- Re: Memory crash on SIP timer re-invite on iOS
- From: Colin Morelli <colin.morelli@xxxxxxxxx>
- Problem with nameservers PJSUA2
- Re: Memory crash on SIP timer re-invite on iOS
- From: Colin Morelli <colin.morelli@xxxxxxxxx>
- Memory crash on SIP timer re-invite on iOS
- From: Colin Morelli <colin.morelli@xxxxxxxxx>
- Re: Assertion failed: (a->addr.sa_family == PJ_AF_INET || a->addr.sa_family == PJ_AF_INET6)
- From: "Fox, Jason P." <jfox@xxxxxxxxxxxxxxx>
- Re: Hard coded telephony-event/8000
- From: JOHAN LANTZ <johan.lantz@xxxxxxxxxxxxxx>
- Re: Error when using PJSUA 2.6 Python module
- From: Viacheslav Mikerov <slavamikerov@xxxxxxxxx>
- Hard coded telephony-event/8000
- From: Oren Barash <oren.elbitsystems@xxxxxxxxx>
- Error when using PJSUA 2.6 Python module
- From: Bogdan Cristea <cristeab@xxxxxxxxx>
- How many threads and what are their purposes?
- From: Viacheslav Mikerov <slavamikerov@xxxxxxxxx>
- When call connects audio is lost
- From: Dingo Egret <sephvelut@xxxxxxxxx>
- android random crash when client call heavily
- From: 高宇 <iorixyz@xxxxxxxxx>
- SIP REC support in PJSUA
- From: Bogdan Cristea <cristeab@xxxxxxxxx>
- Take the Open Source Survey
- From: Perry Ismangil <perry@xxxxxxxxx>
- android random crash when client call heavily
- From: W00HA09.高宇(万睿) <gaoyu01@xxxxxxxxx>
- Re: PBX selection
- From: Roamer2998 <roamer2998@xxxxxxxxx>
- Patch: ice_strans default_candidate selection
- From: Colin Morelli <colin.morelli@xxxxxxxxx>
- Re: tls licence question, possible violation
- From: Adonay Felipe Nogueira <adfeno@xxxxxxxxxxxxxxx>
- Re: iOS Error: "Address already in use"
- From: "qiulang"<qiulang@xxxxxxxxxxx>
- Re: iOS Error: "Address already in use"
- From: Frederik Riedel <frederik.riedel@xxxxxxxx>
- One-Way SRTP Issues
- From: Colin Morelli <colin.morelli@xxxxxxxxx>
- Bug in SRTP negotiation
- From: Colin Morelli <colin.morelli@xxxxxxxxx>
- Re: PBX selection
- From: Roamer2998 <roamer2998@xxxxxxxxx>
- Re: NAT64 ios issue
- From: Ashok Narvaneni <ashoknarvaneni7@xxxxxxxxx>
- Re: NAT64 ios issue
- From: JOHAN LANTZ <johan.lantz@xxxxxxxxxxxxxx>
- Re: PBX selection
- From: Tzafrir Cohen <tzafrir.cohen@xxxxxxxxxx>
- Re: PBX selection
- From: Roamer2998 <roamer2998@xxxxxxxxx>
- Re: NAT64 ios issue
- From: Ashok Narvaneni <ashoknarvaneni7@xxxxxxxxx>
- Android camera device is not free up after Hang up with back camera on
- From: Сергей Митрофанов <goretz.m@xxxxxxxxx>
- Re: PBX selection
- From: David Villasmil Govea <david.villasmil@xxxxxxxxx>
- PBX selection
- From: Roamer2998 <roamer2998@xxxxxxxxx>
- Re: Bug in SRTP when pjsua_acc_config.use_srtp == PJMEDIA_SRTP_OPTIONAL
- From: David Talmage <sip.phone.fan@xxxxxxxxx>
- Re: NAT64 ios issue
- From: Ashok Narvaneni <ashoknarvaneni7@xxxxxxxxx>
- Re: iOS Error: "Address already in use"
- From: Colin Morelli <colin.morelli@xxxxxxxxx>
- Re: Bug in SRTP when pjsua_acc_config.use_srtp == PJMEDIA_SRTP_OPTIONAL
- From: Colin Morelli <colin.morelli@xxxxxxxxx>
- iOS Error: "Address already in use"
- From: Frederik Riedel <frederik.riedel@xxxxxxxx>
- Re: Bug in SRTP when pjsua_acc_config.use_srtp == PJMEDIA_SRTP_OPTIONAL
- From: David Talmage <sip.phone.fan@xxxxxxxxx>
- Re: Bug in SRTP when pjsua_acc_config.use_srtp == PJMEDIA_SRTP_OPTIONAL
- From: JOHAN LANTZ <johan.lantz@xxxxxxxxxxxxxx>
- Record video call
- From: Bogdan Cristea <cristeab@xxxxxxxxx>
- Bug in SRTP when pjsua_acc_config.use_srtp == PJMEDIA_SRTP_OPTIONAL
- From: David Talmage <sip.phone.fan@xxxxxxxxx>
- pjmedia RTP session status flags are overridden
- From: Karl Horvath <k.horvath.at@xxxxxxxxx>
- Antwort: [Newsletter] pjsip_parse_uri() fails with '@' in URI parameter
- From: Zoltan.Toth.ext@xxxxxxxxxxxxxxxxx
- PJSIP Compilation is failing for MIPS as ABI with latest Android NDK toolchain
- From: "Mehul Hirpara" <mehul.hirpara@xxxxxxxxxxxxxxxx>
- Switching capture device and playback device
- From: Matthew Danielson <matt@xxxxxxxxxxxxxx>
- Video calls on Android - which libs are required?
- From: Zarko Coklin <zcoklin@xxxxxxxxxxx>
- pjsip_parse_uri() fails with '@' in URI parameter
- From: Brian White <mscdex@xxxxxxxxxx>
- Re: NAT64 ios issue
- From: Ashok Narvaneni <ashoknarvaneni7@xxxxxxxxx>
- pjsip cannot unregister properly from SIP server
- From: Анцев Александр <a.antsev@xxxxxxxxx>
- Re: NAT64 ios issue
- From: Ashok Narvaneni <ashoknarvaneni7@xxxxxxxxx>
- Re: NAT64 ios issue
- From: Colin Morelli <colin.morelli@xxxxxxxxx>
- Re: NAT64 ios issue
- From: JOHAN LANTZ <johan.lantz@xxxxxxxxxxxxxx>
- Re: NAT64 ios issue
- From: Ashok Narvaneni <ashoknarvaneni7@xxxxxxxxx>
- Re: Custom SIP INFO message parsing
- From: Alin Radut <claudel@xxxxxxxxx>
- Custom SIP INFO message parsing
- From: John Gathm <john.gathm@xxxxxxxxx>
- Re: NAT64 ios issue
- From: JOHAN LANTZ <johan.lantz@xxxxxxxxxxxxxx>
- Re: NAT64 ios issue
- From: Ashok Narvaneni <ashoknarvaneni7@xxxxxxxxx>
- Re: NAT64 ios issue
- From: Colin Morelli <colin.morelli@xxxxxxxxx>
- Re: NAT64 ios issue
- From: Ashok Narvaneni <ashoknarvaneni7@xxxxxxxxx>
- Re: R: Pjsip Recipe for Yocto
- From: Stepan Salenikovich <stepan.salenikovich@xxxxxxxxxxxxxxxxxxxx>
- Re: Assertion failed: (a->addr.sa_family == PJ_AF_INET || a->addr.sa_family == PJ_AF_INET6)
- From: Lüffe, Stefan <SLueffe@xxxxxxxxxxx>
- Re: NAT64 ios issue
- From: Ashok Narvaneni <ashoknarvaneni7@xxxxxxxxx>
- Re: NAT64 ios issue
- From: Colin Morelli <colin.morelli@xxxxxxxxx>
- NAT64 ios issue
- From: Ashok Narvaneni <ashoknarvaneni7@xxxxxxxxx>
- Re: Unable to configure build environment to generate PJSIP lib for ABI=MIPS using Android NDK
- From: "Mehul Hirpara" <mehul.hirpara@xxxxxxxxxxxxxxxx>
- PJSIP making a simple Phone Call
- From: Symeon Mattes <symeon.mattes@xxxxxxxxxxx>
- Unable to configure build environment to generate PJSIP lib for ABI=MIPS using Android NDK
- From: "Mehul Hirpara" <mehul.hirpara@xxxxxxxxxxxxxxxx>
- can't set Expires header value to 4294967295
- From: "Gross, Jeffrey" <jgross03@xxxxxxxxxx>
- Incoming INVITE with SDP but without media descriptions
- From: Piewald Georg <gpi@xxxxxxxxxxxxxxxxxxxx>
- Re: Issue with reinvite on Android
- From: Colin Morelli <colin.morelli@xxxxxxxxx>
- Issue with reinvite on Android
- From: Colin Morelli <colin.morelli@xxxxxxxxx>
- Re: Interesting deadlock bug found causing three threads to deadlock on the PJSUA, UA, and transaction locks.
- From: Ming <ming@xxxxxxxxx>
- Re: Interesting deadlock bug found causing three threads to deadlock on the PJSUA, UA, and transaction locks.
- From: David Richards <david.brian.richards@xxxxxxxxx>
- Re: Interesting deadlock bug found causing three threads to deadlock on the PJSUA, UA, and transaction locks.
- From: Ross Beer <ross.beer@xxxxxxxxxxx>
- R: R: Pjsip Recipe for Yocto
- From: "Lele" <86eldnl@xxxxxxxxx>
- Re: R: Pjsip Recipe for Yocto
- From: Praveen Kumar <praveen.kumar@xxxxxxxxxxxx>
- Re: R: Pjsip Recipe for Yocto
- From: Daniele Elia <86eldnl@xxxxxxxxx>
- Does PJSIP supports (RFC 6035 - SIP package for sending voice quality metrices)?
- From: shubham verma <shubham.verma2k10@xxxxxxxxx>
- Re: R: Pjsip Recipe for Yocto
- From: Praveen Kumar <praveen.kumar@xxxxxxxxxxxx>
- R: Pjsip Recipe for Yocto
- From: "Lele" <86eldnl@xxxxxxxxx>
- Re: Interesting deadlock bug found causing three threads to deadlock on the PJSUA, UA, and transaction locks.
- From: David Richards <david.brian.richards@xxxxxxxxx>
- Re: Interesting deadlock bug found causing three threads to deadlock on the PJSUA, UA, and transaction locks.
- From: Ming <ming@xxxxxxxxx>
- Re: Interesting deadlock bug found causing three threads to deadlock on the PJSUA, UA, and transaction locks.
- From: David Richards <david.brian.richards@xxxxxxxxx>
- (PJNATH) Mystery packets received via Stun socket - identified as from Spotify
- From: Howard Chalkley <hchalkley@xxxxxxxxxxxxxx>
- Re: Pjsip Recipe for Yocto
- From: Praveen Kumar <praveen.kumar@xxxxxxxxxxxx>
- Pjsip Recipe for Yocto
- From: "Lele" <86eldnl@xxxxxxxxx>
- T.38 FAX support in PJSIP
- From: David Talmage <sip.phone.fan@xxxxxxxxx>
- Re: Need Help in compile PJSIP library for IPhone
- From: Alin Radut <claudel@xxxxxxxxx>
- Crash with TURN enabled caused by an apparent race condition
- From: Alin Radut <claudel@xxxxxxxxx>
- Re: Interesting deadlock bug found causing three threads to deadlock on the PJSUA, UA, and transaction locks.
- From: Ming <ming@xxxxxxxxx>
- Need Help in compile PJSIP library for IPhone
- From: Manoj kumar Dixit <manojdixit.dixit8@xxxxxxxxx>
- Re: Deadlock between dlg and tsx
- From: Ming <ming@xxxxxxxxx>
- Re: Interesting deadlock bug found causing three threads to deadlock on the PJSUA, UA, and transaction locks.
- From: David Richards <david.brian.richards@xxxxxxxxx>
- Re: Deadlock between dlg and tsx
- From: Alex Hermann <alex-lists@xxxxxxxxx>
- Audio Call issue
- From: Ashok Narvaneni <ashoknarvaneni7@xxxxxxxxx>
- Re: audio call switch to video call pjsip
- From: Ashok Narvaneni <ashoknarvaneni7@xxxxxxxxx>
- Re: Deadlock between dlg and tsx
- From: Ming <ming@xxxxxxxxx>
- Re: Interesting deadlock bug found causing three threads to deadlock on the PJSUA, UA, and transaction locks.
- From: Ming <ming@xxxxxxxxx>
- Re: audio call switch to video call pjsip
- From: Ashok Narvaneni <ashoknarvaneni7@xxxxxxxxx>
- Re: Pjmedia Write Frame in Seperate Thread
- From: Kiran Bhosale <kbopensource2@xxxxxxxxx>
- Record incoming calls
- From: Bogdan Cristea <cristeab@xxxxxxxxx>
- Re: Pjmedia Write Frame in Seperate Thread
- From: Toni Rutar <buldozer@xxxxxxxxxx>
- Pjmedia Write Frame in Seperate Thread
- From: Kiran Bhosale <kbopensource2@xxxxxxxxx>
- Unable to make video call using pjsip 2.5.5
- From: Srinivasa Raghavan Sundararajan <ssraghavan76@xxxxxxxx>
- Re: Interesting deadlock bug found causing three threads to deadlock on the PJSUA, UA, and transaction locks.
- From: "janu@xxxxxxxxxxx" <janu@xxxxxxxxxxx>
- Re: Interesting deadlock bug found causing three threads to deadlock on the PJSUA, UA, and transaction locks.
- From: Alex Hermann <alex-lists@xxxxxxxxx>
- SIP Package for Voice Quality Reporting(RFC 6035) in PJSIP
- From: shubham verma <shubham.verma2k10@xxxxxxxxx>
- Interesting deadlock bug found causing three threads to deadlock on the PJSUA, UA, and transaction locks.
- From: David Richards <david.brian.richards@xxxxxxxxx>
- Re: Multiple conference calls on MIPS-125 MHz Embedded Linux
- From: Toni Rutar <buldozer@xxxxxxxxxx>
- [android] tongenerator in conference bridge
- From: Andrzej Grajnert <frogersik@xxxxxxxxx>
- Re: Send audio wav only on left device channel
- From: "Wientzek, Adam Thomas" <adam.wientzek@xxxxxx>
- Multiple conference calls on MIPS-125 MHz Embedded Linux
- From: Kiran Bhosale <kbopensource2@xxxxxxxxx>
- pjsip 2.6 for Android missing linux/android_alarm.h in NDK r14
- From: Anthony Alba <ascanio.alba7@xxxxxxxxx>
- Assertion failed: (a->addr.sa_family == PJ_AF_INET || a->addr.sa_family == PJ_AF_INET6)
- From: Vincent Narbot <vincent.narbot@xxxxxxxxxxxxx>
- Send audio wav only on left device channel
- From: "Lele" <86eldnl@xxxxxxxxx>
- Re: audio call switch to video call pjsip
- From: Ranjit Avasarala <ranjit.avasarala@xxxxxxxxx>
- audio call switch to video call pjsip
- From: Ashok Narvaneni <ashoknarvaneni7@xxxxxxxxx>
- Switch from front to back camera
- From: Ashok Narvaneni <ashoknarvaneni7@xxxxxxxxx>
- Re: Asterisk crashs with PJSIP
- From: ➔ Dan ADAGIO <dan@xxxxxxxxxxxxxxxxx>
- tsx and glck not destroyed
- From: Gerard van den Bosch <gvandenbosch@xxxxxxxxx>
- Re: Windows build fixes
- From: Riza Sulistyo <riza@xxxxxxxxx>
- Re: Asterisk crashs with PJSIP
- From: ➔ Dan ADAGIO <dan@xxxxxxxxxxxxxxxxx>
- Re: Asterisk crashs with PJSIP
- From: Ross Beer <ross.beer@xxxxxxxxxxx>
- Re: Asterisk crashs with PJSIP
- From: ➔ Dan ADAGIO <dan@xxxxxxxxxxxxxxxxx>
- Asterisk crashs with PJSIP
- From: ➔ Dan ADAGIO <dan@xxxxxxxxxxxxxxxxx>
- Presence Update
- From: Ashok Narvaneni <ashoknarvaneni7@xxxxxxxxx>
- PJSIP is core dumping on linux when the clock thread is destroyed.
- From: "Ward, David (Proxy)" <David.Ward@xxxxxxxxxxxxx>
- On Side Audio When Only One Party is Using ICE
- From: Nahum Nir <hello.shalom.hi@xxxxxxxxx>
- Deadlock between dlg and tsx
- From: Alex <alex-lists@xxxxxxxxx>
- Re: data exchange with pjsip modules
- Re: data exchange with pjsip modules
- From: Ranjit Avasarala <ranjit.avasarala@xxxxxxxxx>
- data exchange with pjsip modules
- Windows build fixes
- From: Sean Bright <sean.bright@xxxxxxxxx>
- Re: Video call Issue
- From: deng xiaoming <dengxiaoming@xxxxxxxxx>
- Re: Video call Issue
- From: Ashok Narvaneni <ashoknarvaneni7@xxxxxxxxx>
- Re: PJSIP 2.4.5 when using TCP and TLS references different ports in Via and Contact headers - why?
- From: Žarko Coklin <zcoklin@xxxxxxxxxxx>
- Re: Video call Issue
- From: Ranjit Avasarala <ranjit.avasarala@xxxxxxxxx>
- Re: Video call Issue
- From: Lars Olsson <lars.olsson76@xxxxxxxxx>
- Re: Video call Issue
- From: Ashok Narvaneni <ashoknarvaneni7@xxxxxxxxx>
- Re: PJSIP 2.4.5 when using TCP and TLS references different ports in Via and Contact headers - why?
- From: Nenad Milidrag <nmilidrag@xxxxxxxxxx>
- Re: PJSIP 2.4.5 when using TCP and TLS references different ports in Via and Contact headers - why?
- From: Žarko Coklin <zcoklin@xxxxxxxxxxx>
- Re: Video call Issue
- From: David Villasmil Govea <david.villasmil@xxxxxxxxx>
- Re: Video call Issue
- From: Ranjit Avasarala <ranjit.avasarala@xxxxxxxxx>
- How to use hardware encoded H264 from a camera in a Video Call?
- From: Richard Perez <richard@xxxxxxxxxxxxx>
- Video call Issue
- From: Ashok Narvaneni <ashoknarvaneni7@xxxxxxxxx>
- Re: PjSip UDP NAT keepalive
- From: JOHAN LANTZ <johan.lantz@xxxxxxxxxxxxxx>
- PjSip UDP NAT keepalive
- From: Denis <lookingforheads@xxxxxxxxx>
- Re: PJSUA-LIB - cannot get remote sdp
- From: Nenad Milidrag <nmilidrag@xxxxxxxxxx>
- Re: PJSUA-LIB - cannot get remote sdp
- From: JOHAN LANTZ <johan.lantz@xxxxxxxxxxxxxx>
- Re: PJSUA-LIB - cannot get remote sdp
- From: Nenad Milidrag <nmilidrag@xxxxxxxxxx>
- Voice Quality specialist needed
- From: "Ruddy Gbaguidi" <plugworld@xxxxxxxxxx>
- Can someone make a basic sip softphone android + iphone
- From: vidhya sagar dixit <vids.cs@xxxxxxxxx>
- [Patch] In-band DTMF tone end detection bug
- From: "Thomas Janu" <janu@xxxxxxxxxxx>
- Re: How to send AT command to SIP with pjsip?
- From: Guillaume Roguez <guillaume.roguez@xxxxxxxxxxxxxxxxxxxx>
- [ANDROID] Warning with speex and Jitter
- From: "Fanilo Gabaud" <fanilo@xxxxxxxxxx>
- Using DTMF in pjsua
- From: Niels Thole <nt@xxxxxxxxxxxx>
- Re: PJSUA-LIB - cannot get remote sdp
- From: JOHAN LANTZ <johan.lantz@xxxxxxxxxxxxxx>
- PJSUA-LIB - cannot get remote sdp
- From: Nenad Milidrag <nmilidrag@xxxxxxxxxx>
- Re: How to disable Automatic Gain Control
- From: "Ruddy Gbaguidi" <plugworld@xxxxxxxxxx>
- Re: How to disable Automatic Gain Control
- From: Ruddy Gbaguidi <plugworld@xxxxxxxxxx>
- Re: How to disable Automatic Gain Control
- From: JOHAN LANTZ <johan.lantz@xxxxxxxxxxxxxx>
- How to disable Automatic Gain Control
- From: "Ruddy Gbaguidi" <plugworld@xxxxxxxxxx>
- SIP BYE race between two UA with PJSIP 2.5.5 not ending call
- From: Shane Cole <shane.ian.cole@xxxxxxxxx>
- Multiple contacts/AOR
- From: Nahum Nir <hello.shalom.hi@xxxxxxxxx>
- Re: Basic PJSUA question (without sound card)
- From: "Kpama Frederic" <kpamafrederic@xxxxxxxxx>
- INVITE with Authorization header
- From: Andrzej Grajnert <frogersik@xxxxxxxxx>
- Re: RTP Header extension
- From: Лухнов Андрей Олегович <loukhnov@xxxxxxxxxxx>
- Re: Basic PJSUA question (without sound card)
- From: "Kpama Frederic" <kpamafrederic@xxxxxxxxx>
- Building for Blackberry 10
- From: Lorenzo Ferrante <lorenzoferrante90@xxxxxxxxx>
- Where is sndtest
- From: "Ruddy Gbaguidi" <plugworld@xxxxxxxxxx>
- PJSIP does not use TURN as its last resort!
- From: "曹贵林" <guilin.cao@xxxxxxxxx>
- H264 Video question
- From: Jerry Horel <jhorel@xxxxxxxxxxxxxxxxxxxx>
- RTP Header extension
- From: "Lele" <86eldnl@xxxxxxxxx>
- Re: [android] destroy endpoint on call -crash GC
- From: frogersik <frogersik@xxxxxxxxx>
- Re: [android] destroy endpoint on call -crash GC
- From: Michael Barthold <Michael.Barthold@xxxxxxxxxxx>
- [android] destroy endpoint on call -crash GC
- From: frogersik <frogersik@xxxxxxxxx>
- PJSUA Python cannot open sound device
- From: Eric Le Bras <eric.lebras@xxxxxxxxx>
- Is possible to set 2 output audio devices?
- From: "Lele" <86eldnl@xxxxxxxxx>
- Forcing the use of simple GUID generator
- From: <brian@xxxxxxxxxxxxxxxxxxxx>
- PJSIP Thread Query
- From: Sourav Bhowmik <sou.bhowmik02@xxxxxxxxx>
- PJSIP 2.4.5 when using TCP and TLS references different ports in Via and Contact headers - why?
- From: Zarko Coklin <zcoklin@xxxxxxxxxxx>
- Forced memory cleaning
- From: Василий <vasmosh@xxxxxxxxx>
- PJSIP 2.6 IPv6 ICE fails... sometimes
- From: Alex <alex.solis@xxxxxxxxxxxxxx>
- Re: PJSIP not sending Authorization Header on 401
- From: Peter Warrick <peter@xxxxxxxxxxxxxx>
- PJSIP not sending Authorization Header on 401
- From: Peter Warrick <peter@xxxxxxxxxxxxxx>
- PJSIP version 2.6 is released with UWP & WP8.x support
- From: Nanang Izzuddin <nanang@xxxxxxxxx>
- Cant start build for armeabi-v7a
- From: Aleksandar Milenkovic <shark@xxxxxxxxxxxxxx>
- Video is not visible on pjsip
- From: Sree Harsha Malepati <SreeHarsha.Malepati@xxxxxxxxxxxxxxxxxxx>
- PJSUA: purpose of pjsua_acc_add_local() after creating a transport
- From: David Talmage <sip.phone.fan@xxxxxxxxx>
- Remove sendrecv attribute from sdp - order/remove fields in SDP
- From: "Lele" <86eldnl@xxxxxxxxx>
- send ACK request after it receives incoming 2xx response for INVITE
- From: "Lele" <86eldnl@xxxxxxxxx>
- Re: SSL timeout
- From: Alex Solis <alex.solis@xxxxxxxxxxxxxx>
- Re: Send Info Message with defined body
- From: JOHAN LANTZ <johan.lantz@xxxxxxxxxxxxxx>
- Hot audio device change
- Re: Creating multiple TCP listeners in PJSUA2
- Possible ICE check bug
- From: Jonni Rainisto <jonni.rainisto@xxxxxxxxx>
- Send Info Message with defined body
- From: "Lele" <86eldnl@xxxxxxxxx>
- Support for BLF PJSIP/PJSUA RFC 4235?
- From: Jose Solares <jsolares@xxxxxxxxxxx>
- Re: Problem with registration and call over WiMAX
- From: Bojan Tanasijevic <bojan.tanasijevic@xxxxx>
- Re: RFC 2833 DTMF digit detection problem
- From: Eric Le Bras <eric.lebras@xxxxxxxxx>
- SSL timeout
- From: Alex <alex.solis@xxxxxxxxxxxxxx>
- Re: Problem with registration and call over WiMAX
- From: Andrzej Grajnert <frogersik@xxxxxxxxx>
- Trigger Sip Messages during Pjsua Call
- From: "Lele" <86eldnl@xxxxxxxxx>
- RFC 2833 DTMF digit detection problem
- From: Eric Le Bras <eric.lebras@xxxxxxxxx>
- Re: Pjsip video support errors in ios
- From: Yongwen ZHU <zhuyongwen@xxxxxxxxxxxxxxxxx>
- [PATCH] Enable creating memory recorder in pjsua
- From: Jiulong Wang <jiulongw@xxxxxxxxx>
- Problem with registration and call over WiMAX
- From: Bojan Tanasijevic <bojan.tanasijevic@xxxxx>
- Inband DTMF Detection
- From: Jason Stäuble <mokitto@xxxxxxxxxxx>
- Pjsip video support errors in ios
- From: Ashok Narvaneni <ashoknarvaneni7@xxxxxxxxx>
- No sound from capture device when communication via Ethernet - Works properly with Wifi.
- From: "Mysoft Systems" <info@xxxxxxxxxxxxxxxxx>
- Re: Problem binding socket in nat_detect.c
- From: Nanang Izzuddin <nanang@xxxxxxxxx>
- Use Pjsua without proxy server, is possible?
- From: "Lele" <86eldnl@xxxxxxxxx>
- Re: How to send AT command to SIP with pjsip?
- From: Telesonic Telesonic <telesonic@xxxxxxxxx>
- unable to reINVITE call when call is ringing
- From: Bob Voorneveld <bob.voorneveld@xxxxxxxxxxxxxxxx>
- why pjsip 2.5 ios video will randomly can't display.
- From: yu gao <gaoyu01@xxxxxxxxx>
- Wired vs WiFi on Windows UWP-Raspberry PI
- From: "Mysoft Systems" <info@xxxxxxxxxxxxxxxxx>
- Re: How to send AT command to SIP with pjsip?
- From: Guillaume Roguez <guillaume.roguez@xxxxxxxxxxxxxxxxxxxx>
- How to send AT command to SIP with pjsip?
- From: Telesonic Telesonic <telesonic@xxxxxxxxx>
- Problem binding socket in nat_detect.c
- From: Howard Chalkley <hchalkley@xxxxxxxxxxxxxx>
- Re: Creating multiple TCP listeners in PJSUA2
- From: Alain Totouom <alain.totouom@xxxxxx>
- Creating multiple TCP listeners in PJSUA2
- From: Aleksandar Milenkovic <shark@xxxxxxxxxxxxxx>
- Issue with registering new event with evsub.c
- From: Muhammad Usman Shahid <usmanshahidmuhammad@xxxxxxxxx>
- Adding ":" To Costume Contact Field
- From: Nahum Nir <hello.shalom.hi@xxxxxxxxx>
- NetEQ Support?
- From: wjhtinger <wjhtinger@xxxxxxxx>
- call transfer with pjsua2
- Soft Deadlock in pjsua_call_hangup?
- From: Thomas Janu <janu@xxxxxxxxxxx>
- Sound device not closed on call disconnect
- From: sufi al hussaini hassani kamili raheemi <sufialhussaini@xxxxxxxxx>
- Re: pjsip_endpt_unregister_module crash at pjsua_init (pjsip2.5)
- From: "qiulang"<qiulang@xxxxxxxxxxx>
- [bug] Python samples "registration.py" race-condition
- From: Roland Koebler <rk-list@xxxxxxxxxxxxxxxxxxxx>
- [bug] Python pjsua: Error should derive from Exception
- From: Roland Koebler <rk-list@xxxxxxxxxxxxxxxxxxxx>
- capture log-messages of pjsua.Lib()
- From: Roland Koebler <rk-list@xxxxxxxxxxxxxxxxxxxx>
- Re: pjsip android - set max call to 1 makes app crash
- From: Andrzej Grajnert <frogersik@xxxxxxxxx>
- Implementing custom mediaport for PJSUA2 (stdin/stdout)
- From: bloodcarter <bloodcarter@xxxxxxxxx>
- detect worker-thread-exit/fail?
- From: Roland Koebler <rk-list@xxxxxxxxxxxxxxxxxxxx>
- pjsip_endpt_unregister_module crash at pjsua_init (pjsip 2.5)
- From: "qiulang"<qiulang@xxxxxxxxxxx>
- Re: Cannot register with SIP server
- From: Eric Le Bras <eric.lebras@xxxxxxxxx>
- Multiple local accounts
- Re: Cannot register with SIP server
- From: David Villasmil Govea <david.villasmil@xxxxxxxxx>
- Re: Cannot register with SIP server
- From: David Villasmil Govea <david.villasmil@xxxxxxxxx>
- pjsip android - set max call to 1 makes app crash
- From: Andrzej Grajnert <frogersik@xxxxxxxxx>
- Re: Cannot register with SIP server
- From: Eric Le Bras <eric.lebras@xxxxxxxxx>
- Re: Building pjsip with Maven
- From: Ming <ming@xxxxxxxxx>
- Re: Building pjsip with Maven
- From: David Talmage <sip.phone.fan@xxxxxxxxx>
- Re: Cannot register with SIP server
- From: Jason Stäuble <mokitto@xxxxxxxxxxx>
- Detecting DTMF
- From: "Nicolas Dubois" <nicolas.dubois@xxxxxxxxxxxxx>
- Getting 488/Unacceptable Here on external phone calls
- From: Jason Stäuble <mokitto@xxxxxxxxxxx>
- Building pjsip with Maven
- From: David Talmage <sip.phone.fan@xxxxxxxxx>
- Change remote address on the same transport
- From: Muhammad Usman Shahid <usmanshahidmuhammad@xxxxxxxxx>
- unable to build pjsip project for UWP
- From: dragy mil <dragy974@xxxxxxxxx>
- Cannot register with SIP server
- From: Eric Le Bras <eric.lebras@xxxxxxxxx>
- Multiple TCP/TLS Listener
- From: Bernhard Schmidt <berni@xxxxxxxxxxxxx>
- Re: Changing codec parameter dynamically
- Re: Changing codec parameter dynamically
- From: JOHAN LANTZ <johan.lantz@xxxxxxxxxxxxxx>
- Re: Changing codec parameter dynamically
- From: Gang Liu <gangban.lau@xxxxxxxxx>
- Re: Changing codec parameter dynamically
- From: Bill Gardner <billg@xxxxxxxxxxxx>
- Changing codec parameter dynamically
- IPv6 and custom sip transport?
- From: Guillaume Roguez <guillaume.roguez@xxxxxxxxxxxxxxxxxxxx>
- pjsua on Raspberry Pi (SEKCobra - John) - answer by Silvio
- [ANDROID] Speec warning
- From: "Fanilo Gabaud" <fanilo@xxxxxxxxxx>
- Enable OpenGL on linux platform
- From: 藍佳凡 <lan_chia_fan@xxxxxxxxxxxx>
- strtok
- From: ian gilmour <ian.gilmour.x@xxxxxxxxx>
- [ANDROID]Problem loading pjsip with OpenSSL
- From: "Fanilo Gabaud" <fanilo@xxxxxxxxxx>
- Re: Android & Open SSL 1.1
- From: Arslan Pervaiz <arslan.bn@xxxxxxxxx>
- Re: Android & Open SSL 1.1
- From: Arslan Pervaiz <arslan.bn@xxxxxxxxx>
- Re: Android & Open SSL 1.1
- From: "Fanilo Gabaud" <fanilo@xxxxxxxxxx>
- Re: Android & Open SSL 1.1
- From: Arslan Pervaiz <arslan.bn@xxxxxxxxx>
- Android & Open SSL 1.1
- From: "Fanilo Gabaud" <fanilo@xxxxxxxxxx>
- Re: pjsua on Raspberry Pi
- From: Tzafrir Cohen <tzafrir.cohen@xxxxxxxxxx>
- Re: pjsua on Raspberry Pi
- From: SEKCobra <sekcobra@xxxxxxxxx>
- Re: pjsua on Raspberry Pi
- From: SEKCobra <sekcobra@xxxxxxxxx>
- Transport adaptor in pjsua2
- Re: PJSUA2 Audio problems
- pjsua on Raspberry Pi
- From: SEKCobra <sekcobra@xxxxxxxxx>
- Microsip compilation error
- From: Danial Haider <dhaider123@xxxxxxxxx>
- Samples Per Frame Conversion Problem.
- From: Gökhan KILIÇ <gokhan.kilic.89@xxxxxxxxx>
- Re: Android OpenSSL vulnerability issue
- From: JOHAN LANTZ <johan.lantz@xxxxxxxxxxxxxx>
- Android OpenSSL vulnerability issue
- From: Monica Memane <monica.memane@xxxxxxxxxx>
- Re: opencore-amr codecs on Android
- From: JOHAN LANTZ <johan.lantz@xxxxxxxxxxxxxx>
- Re: opencore-amr codecs on Android
- From: Andrzej Grajnert <frogersik@xxxxxxxxx>
- Register packets timeouts
- From: António Richard Silva <silva.antonio@xxxxxxxxx>
- PJSIP and Bitcode
- From: Alin Radut <claudel@xxxxxxxxx>
- Re: Error compiling 5472 on CentOS
- From: Ming <ming@xxxxxxxxx>
- pjsua2 - send unregister before register
- From: Andrzej Grajnert <frogersik@xxxxxxxxx>
- Re: [android] pjsua_create: assertion "status == PJ_SUCCESS" failed'
- From: Andrzej Grajnert <frogersik@xxxxxxxxx>
- snd_clock_rate not working
- From: Muhammed Onur <muhammedonur@xxxxxxxxx>
- Re: [android] pjsua_create: assertion "status == PJ_SUCCESS" failed'
- From: Ming <ming@xxxxxxxxx>
- Re: [android] pjsua_create: assertion "status == PJ_SUCCESS" failed'
- From: Andrzej Grajnert <frogersik@xxxxxxxxx>
- Re: Swift closure isn't allowed to capture context
- From: Alin Radut <claudel@xxxxxxxxx>
- Re: Swift closure isn't allowed to capture context
- From: Colin Morelli <colin.morelli@xxxxxxxxx>
- Swift closure isn't allowed to capture context
- From: Stefan Godoroja <mancunianetz@xxxxxx>
- [pjsua2] Potential memory leak: Dialog seems not be destroyed
- From: "SanveanRu ." <fokin.denis@xxxxxxxxx>
- Re: [android] pjsua_create: assertion "status == PJ_SUCCESS" failed'
- From: Lars Olsson <lars.olsson76@xxxxxxxxx>
- [android] pjsua_create: assertion "status == PJ_SUCCESS" failed'
- From: Andrzej Grajnert <frogersik@xxxxxxxxx>
- [PJSUA][Android] Issue when hanging a call in EARLY state
- From: "Fanilo Gabaud" <fanilo@xxxxxxxxxx>
- Re: pjsip on SoC
- From: H Yavari <hyavari@xxxxxxxxxxxxxx>
- ICE init fails on bad turn server - patch included
- From: Alexandre Viau <aviau@xxxxxxxxxx>
- Re: IPV6 support bug - patch included (now really included)
- From: Alexandre Viau <aviau@xxxxxxxxxx>
- IPV6 support bug - patch included
- From: Alexandre Viau <aviau@xxxxxxxxxx>
- Re: opencore-amr codecs on Android
- From: JOHAN LANTZ <johan.lantz@xxxxxxxxxxxxxx>
- Re: opencore-amr codecs on Android
- From: Andrzej Grajnert <frogersik@xxxxxxxxx>
- Re: Restart ICE Session
- From: cav_345 <cav_345@xxxxxxxxx>
- Restart ICE Session
- From: Colin Morelli <colin.morelli@xxxxxxxxx>
- Bug?: ICE Default Candidate
- From: Colin Morelli <colin.morelli@xxxxxxxxx>
- Error compiling 5472 on CentOS
- From: Ross Beer <ross.beer@xxxxxxxxxxx>
- Re: [PATCH] [RFC] Add attach2 to pjmedia transport interface
- From: b17 c0de <b17c0de@xxxxxxxxx>
- Re: Asterisk segfault in pjsip_endpt_create_pool with PJSIP 2.5.5
- From: Bernhard Schmidt <berni@xxxxxxxxxxxxx>
- Re: IPv6 Media
- From: Colin Morelli <colin.morelli@xxxxxxxxx>
- Re: IPv6 Media
- From: Colin Morelli <colin.morelli@xxxxxxxxx>
- Re: Asterisk segfault in pjsip_endpt_create_pool with PJSIP 2.5.5
- From: Alexei Gradinari <alex2grad@xxxxxxxxx>
- IPv6 Media
- From: Colin Morelli <colin.morelli@xxxxxxxxx>
- Re: Asterisk segfault in pjsip_endpt_create_pool with PJSIP 2.5.5
- From: Bernhard Schmidt <berni@xxxxxxxxxxxxx>
- Re: [PATCH] [RFC] Add pjmedia_stream_get_rtp_sessions() function
- From: Riza Sulistyo <riza@xxxxxxxxx>
- Re: [PATCH] [RFC] Add attach2 to pjmedia transport interface
- From: Riza Sulistyo <riza@xxxxxxxxx>
- Re: Asterisk segfault in pjsip_endpt_create_pool with PJSIP 2.5.5
- From: Joshua Colp <jcolp@xxxxxxxxxx>
- Re: Asterisk segfault in pjsip_endpt_create_pool with PJSIP 2.5.5
- From: George Joseph <gjoseph@xxxxxxxxxx>
- Re: Asterisk segfault in pjsip_endpt_create_pool with PJSIP 2.5.5
- From: Ian Gilmour <ian.gilmour.x@xxxxxxxxx>
- Re: Asterisk segfault in pjsip_endpt_create_pool with PJSIP 2.5.5
- From: Alexei Gradinari <alex2grad@xxxxxxxxx>
- Re: Asterisk segfault in pjsip_endpt_create_pool with PJSIP 2.5.5
- From: Bernhard Schmidt <berni@xxxxxxxxxxxxx>
- Re: Asterisk segfault in pjsip_endpt_create_pool with PJSIP 2.5.5
- From: Ian Gilmour <ian.gilmour.x@xxxxxxxxx>
- Asterisk segfault in pjsip_endpt_create_pool with PJSIP 2.5.5
- From: Bernhard Schmidt <berni@xxxxxxxxxxxxx>
- pjsip on SoC
- From: H Yavari <hyavari@xxxxxxxxxxxxxx>
- Re: PJSUA for SIP to ISDN gateway
- From: Gang Liu <gangban.lau@xxxxxxxxx>
- ios 10 no audio
- From: Bob Voorneveld <bob.voorneveld@xxxxxxxxxxxxxxxx>
- Support James Jones in the Extra Life 2016 event
- From: James Jones <fundraising@xxxxxxxxxxxxxx>
- Re: opencore-amr codecs on Android
- From: Andrzej Grajnert <frogersik@xxxxxxxxx>
- opencore-amr codecs on Android
- From: Andrzej Grajnert <frogersik@xxxxxxxxx>
- Re: opencore-amr codecs on Android
- From: JOHAN LANTZ <johan.lantz@xxxxxxxxxxxxxx>
- opencore-amr codecs on Android
- From: frogersik <frogersik@xxxxxxxxx>
- Re: Android audio device
- From: JOHAN LANTZ <johan.lantz@xxxxxxxxxxxxxx>
- inband DTMF tone generation
- From: frogersik <frogersik@xxxxxxxxx>
- [PATCH] [RFC] Add attach2 to pjmedia transport interface
- From: b17 c0de <b17c0de@xxxxxxxxx>
- [PATCH] [RFC] Add pjmedia_stream_get_rtp_sessions() function
- From: b17 c0de <b17c0de@xxxxxxxxx>
- Android audio device
- From: Lars Olsson <lars.olsson76@xxxxxxxxx>
- Re: [PATCH] [RFC] Add function pjmedia_rtp_decode_rtp2()
- From: Ming <ming@xxxxxxxxx>
- Re: Stop and resume audio with CallKit on iOS 10
- From: Ming <ming@xxxxxxxxx>
- PJ_EINVAL when trying to connect player to sink
- From: "SanveanRu ." <fokin.denis@xxxxxxxxx>
- Re: [PATCH] [RFC] Add function pjmedia_rtp_decode_rtp2()
- From: b17 c0de <b17c0de@xxxxxxxxx>
- Re: [PATCH] [RFC] Add function pjmedia_rtp_decode_rtp2()
- From: b17 c0de <b17c0de@xxxxxxxxx>
- Re: many crashes in iOS10, maybe because to keep-alive timer
- From: "qiulang"<qiulang@xxxxxxxxxxx>
- Stop and resume audio with CallKit on iOS 10
- From: Stephan Swart <stephanswrt@xxxxxxxxx>
- DNS resolver returns errno 97 when IPv6 is disabled
- From: Szőts Ákos <szotsaki@xxxxxxxxx>
- many crashes in iOS10, maybe because to keep-alive timer
- From: "qiulang"<qiulang@xxxxxxxxxxx>
- camera driver splitting
- Re: [PATCH] Fix calling memchr or memcpy with NULL pointer
- From: Nanang Izzuddin <nanang@xxxxxxxxx>
- Re: How to set/send Call-Info header?
- From: "Juergen Edner" <juergen@xxxxxxxxxxx>
- Re: [BUG] Data race with pjmedia_stream stopping media
- From: Nanang Izzuddin <nanang@xxxxxxxxx>
- Re: [PATCH] [RFC] Add function pjmedia_rtp_decode_rtp2()
- From: Nanang Izzuddin <nanang@xxxxxxxxx>
- Re: [PATCH] Fix data race in pj_time_decode()
- From: Nanang Izzuddin <nanang@xxxxxxxxx>
- Re: Error printed when deleting a registered pj::Account
- From: Nanang Izzuddin <nanang@xxxxxxxxx>
- Crash in pjsua_acc_add starting with r5455
- From: Dirar Abu-Saymeh <dirar.abusaymeh@xxxxxxxxxxxxx>
- Re: [PATCH] Fix data race in pj_time_decode()
- From: b17 c0de <b17c0de@xxxxxxxxx>
- Re: patch: crash on using already destroyed ssl socket
- From: Nanang Izzuddin <nanang@xxxxxxxxx>
- Re: patch: crash on using already destroyed ssl socket
- From: Alexei Gradinari <alex2grad@xxxxxxxxx>
- Re: PJSIP iOS audio call issue : Unable to find default audio device (PJMEDIA_EAUD_NODEFDEV) [status=420006]
- From: KANNAN PRASAD <kannanprasad87@xxxxxxxxx>
- PJSIP iOS audio call issue : Unable to find default audio device (PJMEDIA_EAUD_NODEFDEV) [status=420006]
- From: KANNAN PRASAD <kannanprasad87@xxxxxxxxx>
- Re: patch: crash on using already destroyed ssl socket
- From: Alexei Gradinari <alex2grad@xxxxxxxxx>
- Re: patch: crash on using already destroyed ssl socket
- From: George Joseph <gjoseph@xxxxxxxxxx>
- PJSUA for SIP to ISDN gateway
- From: Kiran Bhosale <kbopensource2@xxxxxxxxx>
- Re: patch: crash on using already destroyed ssl socket
- From: Nanang Izzuddin <nanang@xxxxxxxxx>
- Re: patch: crash on using already destroyed ssl socket
- From: Ross Beer <ross.beer@xxxxxxxxxxx>
- Re: patch: crash on using already destroyed ssl socket
- From: George Joseph <gjoseph@xxxxxxxxxx>
- Re: patch: crash on using already destroyed ssl socket
- From: Alexei Gradinari <alex2grad@xxxxxxxxx>
- destroy stream
- From: "Zappasodi Daniele" <D.Zappasodi@xxxxxxxx>
- Re: patch: crash on using already destroyed ssl socket
- From: "Teluu Support (Riza Sulistyo)" <support@xxxxxxxxx>
- Re: patch: crash on using already destroyed ssl socket
- From: Alexei Gradinari <alex2grad@xxxxxxxxx>
- Re: patch: crash on using already destroyed ssl socket
- From: George Joseph <gjoseph@xxxxxxxxxx>
- Re: patch: crash on using already destroyed ssl socket
- From: Riza Sulistyo <riza@xxxxxxxxx>
- Re: patch: crash on using already destroyed ssl socket
- From: Alexei Gradinari <alex2grad@xxxxxxxxx>
- Re: patch: crash on using already destroyed ssl socket
- From: Riza Sulistyo <riza@xxxxxxxxx>
- Re: patch: crash on using already destroyed ssl socket
- From: Alexei Gradinari <alex2grad@xxxxxxxxx>
- Re: patch: crash on using already destroyed ssl socket
- From: Riza Sulistyo <riza@xxxxxxxxx>
- inband DTMF tone generation
- From: Khandker Mahmudur Rahman <mahmudur85@xxxxxxxxx>
- [PATCH] Fix calling memchr or memcpy with NULL pointer
- From: b17 c0de <b17c0de@xxxxxxxxx>
- [BUG] Data race with pjmedia_stream stopping media
- From: b17 c0de <b17c0de@xxxxxxxxx>
- [PATCH] Fix data race in pj_time_decode()
- From: b17 c0de <b17c0de@xxxxxxxxx>
- Re: iOS 10 problems
- From: JOHAN LANTZ <johan.lantz@xxxxxxxxxxxxxx>
- iOS 10 problems
- From: Roger Huang <roger_2010@xxxxxxxxx>
- Re: patch: crash on using already destroyed ssl socket
- From: Ross Beer <ross.beer@xxxxxxxxxxx>
- [PATCH] [RFC] Add function pjmedia_rtp_decode_rtp2()
- From: b17 c0de <b17c0de@xxxxxxxxx>
- Re: Error printed when deleting a registered pj::Account
- From: b17 c0de <b17c0de@xxxxxxxxx>
- Error printed when deleting a registered pj::Account
- From: b17 c0de <b17c0de@xxxxxxxxx>
- Re: [PATCH] [RFC] Add function pjsip_tdata_get_dlg()
- From: b17 c0de <b17c0de@xxxxxxxxx>
- Re: [PATCH] [RFC] Add function pjsip_tdata_get_dlg()
- From: Ming <ming@xxxxxxxxx>
- [PATCH] [RFC] Add function pjsip_tdata_get_dlg()
- From: b17 c0de <b17c0de@xxxxxxxxx>
- Re: [PATCH] Fix leak in alsa_factory_refresh() on "null" device name
- From: Ming <ming@xxxxxxxxx>
- Re: [PATCH] Fix global buffer overflow @sip_msg.c:254
- From: Ming <ming@xxxxxxxxx>
- [PATCH] Fix leak in alsa_factory_refresh() on "null" device name
- From: b17 c0de <b17c0de@xxxxxxxxx>
- [PATCH] Fix global buffer overflow @sip_msg.c:254
- From: b17 c0de <b17c0de@xxxxxxxxx>
- asking for psjua2 support - onNatDetectionComplete
- From: Martin Taschen <martin.taschen@xxxxxxxxx>
- PJSIP Android UnSatisfiedLinkError
- From: Monica Memane <monica.memane@xxxxxxxxxx>
- SRTP: Unable to create media session: No active media stream after negotiation
- From: Анцев Александр <a.antsev@xxxxxxxxx>
- Using OPUS on Android
- From: Scott M Ober <smo@xxxxxxxxxxx>
- Re: Please help - wav_writer writing silence when switch to 1 conference bridge PER call (following pjsua-lib-perf FAQ)
- From: Bill Gardner <billg@xxxxxxxxxxxx>
- Re: Please help - wav_writer writing silence when switch to 1 conference bridge PER call (following pjsua-lib-perf FAQ)
- From: Michael Leonard <emailmleonard@xxxxxxxxx>
- Re: Please help - wav_writer writing silence when switch to 1 conference bridge PER call (following pjsua-lib-perf FAQ)
- From: Bill Gardner <billg@xxxxxxxxxxxx>
- Please help - wav_writer writing silence when switch to 1 conference bridge PER call (following pjsua-lib-perf FAQ)
- From: Michael Leonard <emailmleonard@xxxxxxxxx>
- [pjsua] How to add a custom message header?
- From: Sunny Mok <apollomok@xxxxxxxxxxx>
- Intel IPP
- From: Muhammed Onur <muhammedonur@xxxxxxxxx>
- Re: patch: crash on using already destroyed ssl socket
- From: Ming <ming@xxxxxxxxx>
- Re: patch: crash on using already destroyed ssl socket
- From: Ross Beer <ross.beer@xxxxxxxxxxx>
- Re: pjusa2 - unable to define stunserver
- From: Анцев Александр <a.antsev@xxxxxxxxx>
- pjusa2 - unable to define stunserver
- From: Hermann Norpois <hnorpois@xxxxxxxxx>
- Compiling for iOS with TLS and SRTP support
- From: Matan D <matan@xxxxxxxxxxxxxxxx>
- Re: PJSUA2 Audio problems
- From: Scott M Ober <smo@xxxxxxxxxxx>
- pjsua2 - defining stun server
- From: Hermann Norpois <hnorpois@xxxxxxxxx>
- undefined reference to `pj_dns_parse_addr_response'
- From: Shantala R <shantala.r@xxxxxxxxx>
- PJSUA2 Audio problems
- From: Scott M Ober <smo@xxxxxxxxxxx>
- How to set/send Call-Info header?
- From: "Juergen Edner" <juergen@xxxxxxxxxxx>
- Re: Create account without send request REGISTER
- From: JOHAN LANTZ <johan.lantz@xxxxxxxxxxxxxx>
- Create account without send request REGISTER
- From: Василий <vasmosh@xxxxxxxxx>
- Re: (no subject)
- From: Lokesh Bhat <lokeshbhat1991@xxxxxxxxx>
- pjsip 2.5.5 Issue with video resizing
- From: Alexander Butenko <a.butenka@xxxxxxxxx>
- Re: [Android] Problem incoming call
- From: "Fanilo Gabaud" <fanilo@xxxxxxxxxx>
- Re: [Android] Problem incoming call
- From: Lokesh Bhat <lokeshbhat1991@xxxxxxxxx>
- Re: Patch: don't teminate the session when it doesn't have proper credentials
- From: Yaacov Akiba Slama <ya@xxxxxxxxxxx>
- SIP registration failed - too many hops
- From: Hermann Norpois <hnorpois@xxxxxxxxx>
- Replacing a media transport using on_create_media_transport
- From: Soup <zachary.hueras@xxxxxxxxx>
- Re: (no subject)
- From: Lokesh Bhat <lokeshbhat1991@xxxxxxxxx>
- (no subject)
- From: "Fanilo Gabaud" <fanilo@xxxxxxxxxx>
- (no subject)
- From: Lokesh Bhat <lokeshbhat1991@xxxxxxxxx>
- Register successful for first sip, but while registering second sip application is failing.
- From: Sourav Bhowmik <sou.bhowmik02@xxxxxxxxx>
- patch: crash on using already destroyed ssl socket
- From: Alexei Gradinari <alex2grad@xxxxxxxxx>
- problem to get the address from an hostname
- From: Nicolas Jäger <jagernicolas@xxxxxxxxxx>
- Re: Android Pjsip get AudioRecorder instance
- From: Monica Memane <monica.memane@xxxxxxxxxx>
- Patch: don't teminate the session when it doesn't have proper credentials
- From: Yaacov Akiba Slama <ya@xxxxxxxxxxx>
- Re: Android Pjsip get AudioRecorder instance
- From: Monica Memane <monica.memane@xxxxxxxxxx>
- Re: Linking C-API functions, when using pjsua2
- From: Piewald Georg <gpi@xxxxxxxxxxxxxxxxxxxx>
- Re: Linking C-API functions, when using pjsua2
- From: Walter Doekes <walter+pjsip@xxxxxx>
- Linking C-API functions, when using pjsua2
- From: Piewald Georg <gpi@xxxxxxxxxxxxxxxxxxxx>
- Re: SWIG build error
- From: Scott M Ober <smo@xxxxxxxxxxx>
- SWIG build error
- From: Scott M Ober <smo@xxxxxxxxxxx>
- Re: media.cpp: uncommented code
- From: Piewald Georg <gpi@xxxxxxxxxxxxxxxxxxxx>
- Re: media.cpp: uncommented code
- From: Ming <ming@xxxxxxxxx>
- media.cpp: uncommented code
- From: Piewald Georg <gpi@xxxxxxxxxxxxxxxxxxxx>
- Issues with SRTP and simplex audio
- From: Lars Olsson <lars.olsson76@xxxxxxxxx>
- PJSUA audio latency
- From: sanjay marathe <sanjay_marathe@xxxxxxxxxxx>
- Re: Android Pjsip get AudioRecorder instance
- From: JOHAN LANTZ <johan.lantz@xxxxxxxxxxxxxx>
- Android Pjsip get AudioRecorder instance
- From: Monica Memane <monica.memane@xxxxxxxxxx>
- alsa dependency
- From: Jerry Fath <jaf@xxxxxxx>
- Re: ipv6 uri to be used for calling
- From: varun pratapsingh <varunps2003@xxxxxxxxx>
- ipv6 uri to be used for calling
- From: rohit sethi <rohitksethi@xxxxxxxxx>
- Re: pj_sock_bind sometime failed for ipv6
- From: "qiulang"<qiulang@xxxxxxxxxxx>
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