Follow us on:

Asterisk pjsip codec negotiation

asterisk pjsip codec negotiation asterisk-g72x G. 1 and Certified Asterisk 13. 0. 16. The peer’s phone send the codec list as (uLaw, speex) in 200 OK replay. The PJSIP channel driver enables Asterisk to handle SIP endpoints, such as the phones that you will connect to your Asterisk server. A brief tutorial to provide private voice over IP (VoIP) services using Asterisk in OpenBSD…. C--dis-codec=NAME: Disable codecs with matching NAME. In extended mode MicroSIP will show you, what codec was selected for session. ; * Endpoint "endpoint" ; * Configures core SIP functionality related to SIP endpoints. --snd-clock-rate=HZ PJSIP version 2. 65 64bit installed with Asterisk 13. You can find information about how to handle these issues in the security team's documentation . 4 06 Sep 2019 13:25 minor feature: AST-2019-004 - res_pjsip_t38. Try JIRA - bug tracking software for your team. so with pjsip Settings – Codecs: Eigentlich: g722, alaw Momentan: alaw. I try my best to write down all necessary basic steps. Finally, reload PJsip to allow the above changes to take effect: asterisk -rx "module reload res_pjsip. The media quality also sets speex codec quality/complexity to the number. 1. 100rel - Allow support for RFC3262 provisional ACK tags; aggregate_mwi - Condense MWI notifications into a single NOTIFY. On my current version everything works fine, but after conncting my app to Asterisk 17 it did not receive ChannelHangupRequest event and the following Much of the Asterisk information on the internet is old. 729 but its not free. 2 06 Mar 2021 01:45 minor feature: AST-2021-006 - res_pjsip_t38. com. 13-cert7. core show codec -- Shows a specific codec Show pjproject to Asterisk log mappings pjsip dump endpt -- Dump the res_pjsip endpt internals Linked Applications. 4 Version of this port present on the latest quarterly branch. Colp Fri, 04 Oct 2019 03:07:03 -0700 On Fri, Oct 4, 2019, at 1:45 AM, Andreas Wehrmann wrote: > > On 03/10/2019 16:24, Joshua C. 0 to ports - Update g72x module to 1. LGVOIP_IT*CLI> core show translation paths ilbc--- Translation paths SRC Codec "ilbc" sample rate 8000 --- ilbc:8000 To g723:8000 : No Translation Path Your asterisk server has to be set to accpet speex codec before it can translate it to G711. This is a general package update to the CURRENT release repository based upon TrueOS 19. c: Event presence does not match asterisk-devicestate [2015-02-11 12:28:26] DEBUG[32693] res_pjsip_pubsub. I have also tried compiling and running on our Fedora Core 7 platform, as well as a Debian VM. confirm using pjsip – since chan_sip is depriciate. The result is an easier to understand negotiation process that's implemented in far less code. State of PJSIP in Asterisk 12. conf configuration in the pjsip. Now is the time to migrate users to PJSIP, and there are tools available to help you do Example of codec names: pcma, pcmu, speex/8000, speex/16000, speex/32000, ilbc, gsm, l16/44100/2, etc. 7. We are running Asterisk 13. pjsip. In Asterisk, this is done in the asterisk -r / rasterisk command line interface. 6 is released with UWP & WP8. 27. Other channels may benefit too, as long as they are using RTP. While we did not quite reach full feature parity, the PJSIP stack is feature rich and suitable for many deployment scenarios. 2. 7. 729 Category: Resources/res_pjsip_pubsub ASTERISK-22952: res_pjsip_pubsub: crash when subscription_destructor is terminated from a non-PJSIP thread Revision: 404554 Reporter: mjordan Coders: jcolp ASTERISK-23129: segfault in res_pjsip_pubsub. Asterisk supports the following narrow-band and wideband (HD audio) codecs: G. This list will consist of only those codecs found in both. 25 on centos 6; How to install G729 Codec in Asterisk 13; How to install Alembic and create pjsip tables in asterisk 13. 2, and 16. 1 mit pjsip stack Hardware: 2x QuadCore, 128GB ECC-RAM, 4x HDD Raid 10 Asterisk 11 and previous: chan_sip is the primary option. The assumption throughout this thread is that this is a NAT issue, but codec misconfig can also be a source of 1 way audio. pcap -p -n -s 0 – Dhananjay Kashyap Jun 16 '16 at 9:35 I am looking to replace my older Asterisk 15 VoIP Server with Asterisk 17. « pjsip adds support for OpenCORE AMR-WB AMR-NB codec PJSIP version 2. 9 caused us to revert some of their changes as a work around. g: H26x picture resolution negotiation, this negotiation result will be applied to codec param (for stream in opening the codec) Asterisk Project Security Advisory - AST-2021-005 Posted Feb 19, 2021 Authored by Joshua Colp, Mauri de Souza Meneguzzo | Site asterisk. Distro Stable-6. With Advanced Codec Negotiation that’s about to change! One of the Asterisk team’s goals for 2020 was to dig Use Gerrit: - asterisk/asterisk The stream topology (list of streams and order) is now stored with the configured PJSIP endpoints and used during the negotiation process. log logfile. 722 – 16 kHz wideband codec; passthrough, playback and recording in Asterisk 1. org. 7 is released with DTLS for SRTP keying support, and iOS and Mac native H. org [mailto:pjsip-***@lists. and uses bandwidth donated to the open source Asterisk community by API Digital Communications in Huntsville, AL USA. conf context will be set to default the following line should be added to [default] section. 1 and Certified Asterisk 13. 0. 1, 14 before 14. 08 Now Available. I understand that these patches will make it in to asterisk 1. I have an Asterisk server at home I built few years ago. Codec negotiation in VoipNow. Enable the Internet Low Bitrate Codec (iLBC) ldap: (pjsip) portaudio: Add support for the crossplatform Understanding codec negotiation Applies to VoipNow Professional, VoipNow 3, VoipNow 3. signaling, media features, and NAT traversal, among other things that have been taken care of by PJSIP. The two configuration files that will be dealt with in setup are sip. 0 * app_mixmonitor now sends manager events MixMonitorStart, MixMonitorStop and MixMonitorMute when the channel monitoring is started, stopped and muted (or unmuted) respectively. Category: Channels/chan_sip/General Try using TCP and enable notice in logger. Asterisk 18 отнесён к категории выпусков с расширенной поддержкой (LTS), обновления для которого будут выпускаться в течение пяти лет вместо свойственных для обычных выпусков двух лет. I successfully build pjproject-2. 7. 2 and 16. 1872 * A new SIP channel driver for Asterisk, chan_pjsip is built on the PJSIP. This option maybe specified multiple times. Setup Asterisk with a webphone extension Configure an extension exactly the same way as you do for other endpoints such as a softphone. pjsua defaults to 20 ms. Format attribute. x through 16. > > > If I don't, pjsua encodes at 30 ms and decodes at 20 ms, even though > Asterisk thinks both directions are using 30 ms. An issue was discovered in Asterisk Open Source 13 before 13. 7. I stand corrected about libs installed by asterisk-pjsip. Everything seemed to be working fine for a couple of days and we now get crashes more and more. 0 that used in it. 4 trunk soon 15. digium. Time to start coding? 11. Port details: asterisk15 Open Source PBX and telephony toolkit 15. 1, 14 before 14. changes of Package asterisk----- Fri Mar 26 12:25:27 UTC 2021 - Michael Ströder <michael@stroeder. 264 VideoToolbox codec PJSIP version 2. Now you should be able to go back to your OBi How to configure a Digium SIP Trunking account with Asterisk using chan_pjsip Depending on the version of Asterisk that you are using, You may have two channels drivers that you could use in order to create a peer that you could use to place and receive calls, if you are looking for how to configure asterisk with chan_sip we have another KB article that talks about the configuration. 7. where both call legs use the same codec and Asterisk does not need to translate. org Port Added: 2015-02-26 13:19:15 Last Update: 2020-10-23 09:13:58 SVN Revision: 553096 License: not specified in port Description: G. I’m using FreePBX 14. I imaging this would have to be done in the dialplan, and I've seen posts about codec-related dialplan variables, but it's hard to get any details. atlanta. e. x through 15. While here use DIST_SUBDIR to keep all the asterisk files in one subdirectory. Summary [Back to Top] This release is a point release of an existing major version. 3 x86_64 with pjsip as pulled from the Asterisk github without issue. 1369 * Added format attribute negotiation for the Opus codec. so MD5 digest dialplan functions 0 Running core func_pitchshift. I needed an auto dialer for my CUCM 11. I just tested this new firmware on a T46g and can attest it fixes the codec issue described in this thread when using PJSIP on Asterisk. View Analysis Description Analysis Description The configuration in pjsip. 20 at 14:00 Asterisk telephony/asterisk > > I just tested the new codec negotiation feature and unfortunately wasn't > able to get it add codec specific negotiation, perhaps sdp_neg. PJSIP seems to be more powerful, but use the standard SIP module for this setup. 0-rc2 Reported by: Nic Colledge [32fc784284] Alexei Gradinari License #5691 -- res_sorcery_realtime: Fix regex regression. Not only does this create new configuration opportunities but also completely refactors the negotiation process itself. res_pjsip: Adjust outgoing offer call pref. 164 with 8 digit alternate numbe The credits go to this guy for installing Asterisk & PJSIP. check out what you configured in sip settings and on the trunk compared to what the ITSP or whomever is sending you the call expects wrt to codec This is usually due to either Codec Settings, or NAT configuration. codec_mgr = pjmedia_endpt_get_codec_mgr * This function is called whenever SDP negotiation has completed: 265 File asterisk. I set up a AsteriskNow 1. com is the number one paste tool since 2002. 8. {quote} > > As an aside, chan_pjsip has an analogous dialplan function "PJSIP_MEDIA_OFFER". 18. To start, Asterisk needs a base config for PJSIP at /etc/asterisk/pjsip. 0. x support PJSIP version 2. T… 2: 223: March 4, 2021 I understand at a basic level how codec selection works in Asterisk, so I'm not too hopeful, but I thought I'd throw it out there in case anybody had come up with a solution to this. You can choose between two SIP stacks in Asterisk: chan_sip and chan_pjsip. level 1 encryption=yes ; Tell Asterisk to use encryption for this peer: avpf=yes ; Tell Asterisk to use AVPF for this peer: icesupport=yes ; Tell Asterisk to use ICE for this peer: context=sip. atlanta. Note that SIP codec negotiation is not really negotiation (both sides say what they will accept), but Asterisk will not respond with a codec if it hasn't been offered, and won't use one in the RTP if it wasn't in common between the two sides. We are using free-pbx as a “telephone-board” for a non-profit, all volunteer internet radio station. org/wiki/display/AST/PJSIP+Advanced+Codec+Negotiation) appears to ultimately be what we’re after, but we’re not comfortable running Asterisk 18 in production just yet. 1. issues. 08. It’s been around since the beginning and over the past two decades it’s grown and mutated into one of the least understood parts of Asterisk. 4. This is a very simple SIP User Agent application that only use PJSIP (without PJSIP-UA). soho-piper. 711 codecs from the build –disable-l16-codec Exclude Linear/L16 codec family from the build –disable-gsm-codec Exclude GSM codec in the build –disable-g722-codec Exclude G. It means that the functionality has not yet been exposed in PJSIP yet. I suspected a codec mismatch. pjsip video guide, Aug 15, 2019 · Project Trident 19. 6 is released with UWP & WP8. Confirm that yo are not using Usually in Asterisk PJSIP it can happen due to two things. It is the standard for WebRTC because of how resilient it is in bad network conditions. 8-cert7. com> Thu, 27 Dec 2018 [ 15:55 madpilot] 488546 net/asterisk13/Makefile 488546 net/asterisk13/distinfo 488546 net/asterisk15/Makefile A: Voice quality depends on audio codec that was selected in negotiation for current call session. org. 7. 21-cert3. conf 내용 정리. Frame rate is not critical (between 1 and 5 fps). It works quite well. c because it is handled in a similar manner in later versions of Asterisk. Given a scenario where an outgoing call is placed from Asterisk to a remote SIP server it is possible for a crash to occur. 0 -> Asterisk 13. > I can get the two to communicate if I tell pjsua to use 30 ms. The current Asterisk LTS version is 13 and it come with support of PJSIP. 3_3 net =0 1. In order to allow user control of codec order at those positions Asterisk defines the following PJSIP endpoint options: 1. 4. Raspberry pi install. even i need the samething i. Not all, but most:-rwxr-xr-x 1 root root 96032 May 10 02:06 libpj. Currently I have an application that works with Asterisk over ARI (i. c:856 chan_pjsip_write: Channel PJSIP/anonymous-00000063 asked to send alaw frame when native formats are (g723) (rd:g723->g723; wr:slin->alaw;(slin@8000)->(alaw@8000 An issue was discovered in Asterisk Open Source through 13. The first goal for PJSIP in Asterisk 12 was to strive for feature parity with the existing SIP channel driver. Return: (capture_dev_id, playback_dev_id) tuple handle_events(self, timeout=50) Poll the events from underlying pjsua library. I use a confbridge and in-studio softphone to bridge any phone callers tot he live studio sound board. 18. 08. Go to the directory where the configuration files are located: cd /etc/asterisk Configure a Web SIP channel for Asterisk 11 and previous You need to use chan_sip. 29. 0. Like this: MAKE A BACKUP before trying Asterisk 13 - I made the mistake of trying to use it on a system that runs on a virtual machine and found that getting SIP connections to work could be VERY tricky. The additional advanced codec negotiation options have also been removed from the sample configuration and marked as reserved for future functionality in XML documentation. Colp wrote: > > In PJSIP there is the PJSIP_MEDIA_OFFER dialplan function[1] but ultimately > > codec negotiation is not written Provider A is sending (G729, Alaw, uLaw) offer and asterisk dial the peer with its preferred codec order(G729, aLaw, uLaw). 4. 13 with PJSIP Manuais na Lojamundi. conf produced… [101] type=endpoint aors=101 auth=101-auth allow=g722 disallow=all context=from-internal callerid=device <101> dtmf_mode=rfc4733 use_avpf=no ice_support=no media_use_received_transport=no trust_id_inbound=yes send_pai=yes rtp_symmetric=yes rewrite_contact=yes Is the order of allow/disallow Asterisk 의 pjsip 모듈 설정파일 pjsip. Keyword arguments: name -- codec name. Codecs by quality: High quality: [email protected], [email protected],32kHz, [email protected],24kHz, [email protected] View diff against: View revision: Last change on this file since 23613 was 23613, checked in by BrainSlayer, 7 years ago; replace asterisk with latest version Good day, All! I have problem with view video of interlocutor during call. Hintergrund: in meinen Tests schaffte es die Fritz!Box und Asterisk nicht sich auf den alaw Codec zu einigen wenn der Rufaufbau mit g722 begonnen wurde (was passiert wenn dieser in der Liste ganz oben steht), die angewählte Gegenstelle aber dann doch nur alaw konnte. 12. so Mathematical dialplan function 0 Running core func_md5. ASTERISK-27936 - res_pjsip_session doesn't update media when a 200 comes in with a different port than a 183 ASTERISK-26987 - pbx_dundi: Asterisk crashes when unloading module pbx_dundi. com is the number one paste tool since 2002. ; PJSIP Configuration Samples and Quick Reference 2; 3; This file has several very basic configuration examples, to serve as a quick 4; reference to jog your memory when you need to write up a new configuration. 7. 3, 18. c: Event presence does not match asterisk-mwi [2015-02-11 12:28:26] WARNING[32693] res_pjsip_pubsub. conf file concerning an identify object; they come from the code FreePBX generates and are apparently benign. so Get information about a What are all the packages for asterisk or depenedencies? Asterisk PHP with FastAGI; Step by step Installation of Asterisk 13. Next in your dialplan (/etc/asterisk/extensions. 4 is available, or use this more up-to-date patch In PJSIP there is the PJSIP_MEDIA_OFFER dialplan function[1] but ultimately codec negotiation is not written or implemented in the way you need. At the same time, an audio stream cannot be established between two parties unless a mutually supported codec is found and used. PR: 234547 Submitted by: Ludovic Desweemer <ludovic. Add OPUS option to asterisk ports to enable the opus codec. I have couple SIP trunks to severial ITSP, including Google Voice. voip. conf and extensions. \ > While this doesn't allow for setting multiple codecs, it does handle multiple media \ > types, as you can specify both video or audio for the codec i tried to connected asterisk 13 with cisco phone 79XXX using pjsip and stack registering can someone helping me i already talk with people forums and explained me this is a misconfiguration xml but before i has setup that with asterisk 11 using patch ciscocallmanager for that version asterisk 13 replace asterisk with latest version. c should have external format negotiation mechanism to handle format specific negotiation, e. These were tested with jssip on asterisk v17 with res_pjsip. The SIP signalling also passes through Kamailio. 4. It’s been around since the beginning and over the past two decades it’s grown and mutated into one of the least understood parts of Asterisk. conf for the SIP trunks and extensions. We suggest using PJSIP as an upgrade from Chan_SIP, as Chan_SIP is outdated, and the majority of users are moving to PJSIP which provides a number of more The following issues are resolved in this release: Security bugs fixed in this release: ----- * ASTERISK-29057 - pjsip: Crash on call rejection during high load (Reported by Sandro Gauci) Improvements made in this release: ----- * ASTERISK-29055 - Create a Bridge with video_single mode (Reported by sungtae kim) * ASTERISK-29056 - Increase reg Rainer Piper - Bonn - 0228 97167161 or SIP-URI: sip:7000@sip. 8 is released with WebRTC interopability - RTP/SAVPF - SSRC and OPUS param on the fly; PJSIP version 2. Extension Requirement. The available releases are released as versions 16. incoming_call_offer_pref. c: No pjsip - драйвер канала sip в asterisk 12. so-rwxr-xr-x 1 root root 96032 May 10 02:06 libpj. 0. 27. It's able to make and receive call, and play media to the sound device. after a few hours or minutes it s Attached is a patch for \ > 11. org runs on a server provided by Digium, Inc. 7221 codec in the build Asterisk Project Security Advisory - AST-2021-005 Product Asterisk Summary Remote Crash Vulnerability in PJSIP channel driver Nature of Advisory Denial of Service Susceptibility Remote Unauthenticated Sessions Severity Moderate Exploits Known No Reported On December 4, 2020 Reported By Mauri de Souza Meneguzzo (3CPlus) Posted On February 8, 2021 Last Updated On February 8, 2021 Advisory Howdy, I did an installation yesterday of Asterisk 12 beta 2 using Ubuntu 12. 16. Supported options are those fields on the endpoint object in pjsip. After receiving an incoming offer create a list of preferred codecs between those received in the SDP offered by Alice, and Alice's endpoint configuration. Home » Asterisk Users » PJSIP OPTIONS. 2 as Sip Proxy Server. Pastebin is a website where you can store text online for a set period of time. The same thing happens in both cases. conf. The changes included were made to address problems that have been identified in this release series, or are minor, backwards compatible new features or improvements. ASTERISK-26605 - codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded. Asterisk 12 and beyond: You'll probably want to use chan_pjsip (the newest driver), but you still have the option of using chan_sip as well. this is particularly important when using such small devices as wrg54g(s). asterisk. Here is a working pjsip. 13. c: Add NULL checks before using session media After receiving a 200 OK with a declined stream in response to a T. More Robust SIP authentication PJSIP now supports responding to authentication challenge for any realms, by specifying wildcard (“*”) as the realm in the credential (ticket #231 ). This base configuration, taken directly from the sample config, is just enough for PJSIP to listen on the standard UDP port 5060 for SIP. 0 from asterisk-13. allow - Media Codec(s) to allow; codec_prefs_incoming_offer - Codec negotiation prefs for incoming offers. It’s been around since the beginning and over the past two decades it’s grown and mutated into one of the least understood parts of Asterisk. asterisk. Kudos for Yealink for fixing it! Although the issue persisted for many months, Yealink is the first brand affected by this bug that has actually issued a fix. The codec negotiation project itself became more complicated than expected so to meet the Asterisk 18 core freeze we wanted to focus on only the core foundational aspects - making sure those were correct - as changing those are not easy in a release branch. Loading… Dashboards Jun 06, 2018 · I've spent quite some time configuring Asterisk on my VGV7510KW22 and I want to share my configuration in case it might be useful for somebody. 2, and 16. com s= c=IN IP4 host. Config for 3905 SIP Phone: Finally i have managed to dial right extension, but call from cisco cme -> asterisk 13 (pjsip-trunk-g711ulaw) not Question: I want to send video in high resolution between 2 PJSIP-clients (throw Asterisk SIP server or peer-to-peer). If you are using chan_pjsip, rather use Asterisk 16, the guide is exactly the same. so. 16. e. But if the call is rejected, and Asterisk returns to the playback, call drops, and the asterisk says unable to make the negotiation of codecs. The logic was moved to res_pjsip_session. (Reported by Alexander Traud) * ASTERISK-26673 - chan_pjsip: Crash when using CHANNEL dialplan function around masquerade (Reported by Joshua Colp) * ASTERISK-26684 - res_pjsip: Various issues with compact SIP headers (Reported by Joshua Elson) * ASTERISK-26655 - [patch]pjsip View diff against: View revision: Last change on this file since 23613 was 23613, checked in by BrainSlayer, 7 years ago; replace asterisk with latest version I have tested this using PJSIP extensions with a PJSIP trunk all behind the same NAT router with no issues. As PBX we have Asterisk at Audio Options: --add-codec=name Manually add codec (default is to enable all) --dis-codec=name Disable codec (can be specified multiple times) --clock-rate=N Override conference bridge clock rate --snd-clock-rate=N Override sound device clock rate --stereo Audio device and conference bridge opened in stereo mode --null-audio Use NULL audio Asterisk 11. ASTERISK-25702: PjSip realtime DB and Cache Errors since upgrade to asterisk-13. 38 re-invite. c: Fix deadlock loading realtime configuration. There is a pjsip 0. 0-alpha2. You'll see the pjsip. Pjproject returns the dialog locked and with a reference. e pjsip only for sip messaging but its really difficult( my exp) still im not able to separate it correctly. PJSIP Build For Android with Integration of G729 Codec This article would teach you how to build PJSIP libraries for android. PJSIP is the newer and more modern implementation and is the default one. example. codec_opus - This is not a drill - Digium ’ s legal department has cleared a release of an opus codec - Released along with the 14 version of Asterisk and includes a format module and a format attribute module - codec_opus itself is a binary codec that reports back to a stats server the high use count - [ASTERISK-26822 ] - pjsip/cli_commands: pjsip show channelstats shows wrong codec (Reported by Kevin Harwell) - [ASTERISK-26353 ] - res_musiconhold: musiconhold seems to think that the general section is a class and issues warning (Reported by Jonathan Harris) - [ASTERISK-26685 ] - res_pjsip: Crash when using IPv6 and Transport ws,wss Transport between client and webrtc2sip is WSS. This changes the outgoing offer call preference default option to match the behavior of previous versions of Asterisk. 2, FreePBX 13. 188. There are some hints provided internally for outgoing legs but the result is still ultimately independent. 38 initiated re-invite Asterisk would crash when attempting to dereference a NULL session media object. x and 15. 164 Number Asterisk pjsip sip message Freeswitch codec negotiation Four Mistakes for CIOs to Avoid in 2020, or Any Other Aruba Combines Several Components in New SD-Branch Why AMD Had Such an Impressive CES Showing About: In this guide you will find detailed instructions about WebRTC setup for Asterisk 13. in pjproject 2. desweemer@gmail. E-Learning • Asterisk requires the following packages – Asterisk • Optionals – dahdi-linux – drivers das placas – dahdi-tools – utilitários para as placas In this case a very useful item to add to your toolbox is pjsip, which is a SIP stack library (used also by Asterisk and chan_pjsip being the current recommended SIP channel, as opposed to the older chan_sip) that exposes an API and also a command-line option (pjsua). Re: [asterisk-users] Realtime PJSIP max_streams' issues Joshua C. To give you some idea of just how difficult a job this is, a simple call from Alice to Bob currently causes 8 attempts to reconcile codecs between them in app_dial, chan_pjsip, res_pjsip This changes the outgoing offer call preference default option to match the behavior of previous versions of Asterisk. 722 codec in the build –disable-g7221-codec Exclude G. Venha Conferir! Asterisk Inbound Route Config File Configuring the MP-112 requires first configuring it to register as a SIP extension, and then configuring it to send calls to the Asterisk server (as the “proxy”), and then disabling fax detection so that fax calls go through as regular voice calls. Codec Support One is codecs support, make sure you have specified codecs to be used and both sides can communicate on at least on available codec. 5 cluster. PJSIP will not automatically switch the sending one to the receiving one. 2. Monitor the SDP on Asterisk & MediaBroker to see if the problem is related to Codec negotiation. I'm using self-signed certs in Asterisk and webrtc2sip but that seems to work fine after allowing Chrome to add the exception. Pastebin. 4; full support incl. 0 Content-Length: 0 And the debug logging is reporting: [2015-02-11 12:28:26] DEBUG[32693] res_pjsip_pubsub. # asterisk -rx "module show like crypto" Module Description Use Count Status Support Level res_crypto. Asterisk is usually able to translate codecs (so-called transcoding) if the two call legs want to use different codecs, but it is generally preferable to let it operate in pass-through mode – i. Follow the instructions below for the channel driver you chose. You can find information about how to handle these issues in the security team's documentation . With Advanced Codec Negotiation that’s about to change! One of the Asterisk team’s goals for 2020 was to dig PJSIP trunk configurations are filtering video CODECs ( H264 , VP8 etc) outbound. Reboot will work again. Michael Smith wrote: > Hi, > > Asterisk uses a fixed iLBC packetization of 30 ms. 729 codec for Asterisk PBX 1. 0 beta) I set resolution 1680x1050, bitrate 1000, frame rate 3 fps. This works pretty good, but because of the double encoding/decoding using basic G711u codec (one from gvoice motif to PBX and one from PBX to [ASTERISK-28777] – Codec Negotiation: add outgoing_call_offer_prefs option (Reported by Kevin Harwell) [ASTERISK-27946] – dial (API): Storage of dialed target uses AST_MAX_EXTENSION when it shouldn’t (Reported by Joshua Elson) [ASTERISK-28782] – Add support for Content-Disposition header in multi-part INVITES (Reported by Torrey Searle) We’ve been working hard on new codec negotiation stuff for Asterisk 18 and we’ve got some stuff to run by you. 5 and newer! Overview A successful codec negotiation is an essential condition for a successful call between two parties. The you have a codec mismatch , core show translations would be helpful. 10 is released with VP8 and VP9 video codec support little upgrade: from asterisk CLI . 0 from asterisk-13. [ASTERISK-24779] - Passthrough OPUS codec not working with chan_pjsip [ASTERISK-24958] - Forwarding loop detection inhibits certain desirable scenarios [ASTERISK-25116] - res_pjsip: Two PeerStatus AMI messages are sent for every status change [ASTERISK-25135] - [patch]RTP Timeout hangup cause code missing To connect video based webrtc endpoints ensure you load the codecs and also libsrtp . 711 ulaw (as used in US); G. 3 * 4 * Copyright (C) 2013, Digium, Inc. txt file, I have the following codec order for 6001 and 6002: The first thing that you need to configure to deploy the topology is the PJSIP channel driver. Asterisk Project Security Advisory - AST-2021-005 Posted Feb 19, 2021 Authored by Joshua Colp, Mauri de Souza Meneguzzo | Site asterisk. I used a Raspbian light image, but any distro will do. Also capture tcpdump and check on wireshark where any voice packets is being generated or not. AST-2020-001 - res_pjsip: Return dialog locked and referenced. 7. Maintainer: madpilot@FreeBSD. One-way audio for incoming calls - codec negotiation prb? by adron » Mon Apr 16, 2012 8:31 am My problem is that for incoming calls, I can hear the other person, but the other person cannot hear me. 4. 0, 14. Introduction With the release of Asterisk 18 comes a new Advanced Codec Negotiation process. 7-dev python-daemon python-lockfile libv4l-dev libx264-dev libssl-dev libasound2-dev asterisk PJSIP install The cordless has an 8th CODEC, OPUS, and it's the last one in the cordless list. conf . g. 0, 14. Note: chan_sip works fine on Asterisk 13, but chan_pjsip is rather broken. Overwrite the selective conf in this folders with the existing conf of asterisk to run a basic webrtc video call . rtcp_mux. opkg install asterisk15-res-rtp-asterisk opkg install asterisk15-res-rtp-multicast opkg install asterisk15-res-smdi opkg install asterisk15-res-sorcery opkg install asterisk15-res-speech opkg install asterisk15-sounds opkg install asterisk15-voicemail #opkg install asterisk15-res-hep #opkg install asterisk15-res-hep-pjsip Re: [asterisk-users] ConfBridge and sound prompts Joshua C. 21-cert3. so" Don't be surprised if the above reload command produces a few errors from the pjsip. c: Check for session_media on reinvite. 1, and 15 before 15. 0. This page and its sub-pages are intended to help an administrator configure the new SIP resources and channel driver included with Asterisk 12. c: Check for session_media on reinvite. 1, and 15 before 15. (ATA FXS 7000778 > ASTERISK PASS-THRU > CISCO GATEWAY E1 TRUNK > PSTN) If I make a call from the ATA to a Cisco PSTN number, the call works, and is negotiated the g729 codec. conf) there should be the line that will force asterisk to dial an extension. Server: Asterisk PBX 13. Default: 5 (PJSUA_DEFAULT_CODEC_QUALITY). Asterisk 13. The res_pjsip module handles configuration, so we'll mostly speak in terms of configuring res_pjsip. Colp; Re: [asterisk-users] Realtime PJSIP max_streams' issues Dan Cropp; Re: [asterisk-users] Realtime PJSIP max_streams' issues Ahmed Chohan 1 2 days ago · Asterisk Project Security Advisory - AST-2021-001 Product Asterisk Summary Remote crash in res_pjsip_diversion Nature of Advisory Denial of service Susceptibility Remote authenticated sessions Severity Moderate Exploits Known No Reported On December 28 2020 Reported By Ivan Poddubny Posted On January 04 2021 Last Updated On January 04 2021 Advisory Contact gjoseph AT sangoma Asterisk pjsip. I can successfully call into a room and can call out from the room, but there is no audio. conf. My SIP provider supports G711 (alaw) G723 and G739. get_snd_dev(self) Get the device IDs of current sound devices used by pjsua. --clock-rate=HZ: Set the clock rate of the conference bridge. 3, which add support for asterisk 16 - Add asterisk16 flavor and conflicts to asterisk modules ports which support it - Add conflicts to other asterisk versions ports - Add deprecation notice to asterisk15 which will reach EOL on 2019-10-03 - Fix wording on SOUNDS option description * ASTERISK-27039 - chan_pjsip: Device state is idle when channel from endpoint is in early media (Reported by Joshua Colp) * ASTERISK-26996 - chan_pjsip: Flipping between codecs (Reported by Michael Maier) * ASTERISK-26281 - chan_pjsip would send INVITE to 'Unreachable' endpoints (Reported by Jacek Konieczny) get_codec_parameter(self, name) Get codec parameter for the specified codec. It combines signaling protocol (SIP) with rich multimedia framework and NAT traversal functionality into high level API that is portable and suitable for almost any type of systems ranging from desktops, embedded systems, to An issue was discovered in Asterisk Open Source 13 before 13. Что такое pjsip pjsip мультимедийная библиотека с открытым кодом, для реализации протоколов sip, sdp, rtp, stun, turn и ice. From: pjsip-***@lists. Pastebin. for current PJSIP build (2. A pointer dereference in chan_sip while handling SDP negotiation allows an attacker to crash Asterisk when handling an SDP answer to an outgoing T. 3. It is the Asterisk SIP channel driver that should improve the clarity of the calls. 1 18 Oct 2019 00:45 minor feature: Pjproject_bundled: Replace earlier reverts with official. Basic; Overview of Configuration Section Types Used in the Examples ; ; * Transport "transport" ; * Configures res_pjsip transport layer interaction. 6, a backport for 1. 3, which add support for asterisk 16 - Add asterisk16 flavor and conflicts to asterisk modules ports which support it - Add conflicts to other asterisk versions ports - Add deprecation notice to asterisk15 which will reach EOL on 2019-10-03 - Fix wording on SOUNDS option description PJSIP version 2. With the release of Asterisk 13 chan_sip was marked as extended support module, which means that it doesn't receive core support anymore. –disable-g711-codec Exclude G. With this option enabled, Asterisk will attempt to negotiate the use of the "rtcp-mux" attribute on all media streams. An example of pjsip. 0: Pjsip: Unnecessary 603 Decline Because Of Wrong Codec Decision Looking For The Carrier That Owns A Particular DID >> 2 thoughts on - Pjsip Insecure=port,invite Joshua Colp says: Apr 24, 2020 Asterisk Project Security Advisory - The RTP support in Asterisk maintains its own registry of dynamic codecs and desired payload numbers. ASTERISK-26516 - pjsip: Memory corruption with possible memory leak. I see no messages in the log file that seems to help. pcap. For inband dtmf check out this algorithm. In PJSIP there is the PJSIP_MEDIA_OFFER dialplan function[1] but ultimately codec negotiation is not written or implemented in the way you need. asterisk-dev@lists. Is there no way to mimic functionality we previously had in chan_sip? From an Asterisk perspective “core show channel” will also show you what is currently flowing. 4 PJSIP in a production environment and get random crashes. 9. How to Capture and Debug SIP Packets from asterisk using tcpdump and Wireshark : tcpdump -w /tmp/capture-asterisk. The ‘PJSIP Advanced Codec Negotiation’ document (https://wiki. 8. coded) * - proper SDP negotiation * - PCMA/PCMU The patches improve the SIP codec negotiation, ensuring that if possible that transcoding is avoided. A memory leak occurs when an Asterisk pjsip session object is created and that call gets rejected before the session itself is fully established. The transport between webrtc2sip is udp. With Advanced Codec Negotiation that’s about to change! One of the Asterisk team’s goals for 2020 was to dig into the black art of codec negotiation, understand how it’s currently implemented, create a plan for improvement, get the improvements into Asterisk 18, and finally, document the results. 2 06 Mar 2021 01:45 minor feature: AST-2021-006 - res_pjsip_t38. conf. In the case of a call from the internal (private) network to the outside (public) network, the flow of the SIP signalling is as follows: Internal caller >> Kamailio >> Asterisk [Offer] v=0 o=alice 2890844526 2890844526 IN IP4 host. 7 is released with DTLS for SRTP keying support, and iOS and Mac native H. The user * agent should do a proper SDP negotiation and start RTP media once * SDP negotiation has completed. 1. Source install Debian 8 apt-get update Asterisk pjsip Asterisk pjsip Asterisk modifies the origin line in the SDP, and responds with only the preferred codec configured via the dialplan/configuration file Offer Negotiation - Preferred Codec List Alice's phone offers a set of codecs in an INVITE request, where all codecs are supported by Alice's endpoint The Asterisk Development Team would like to announce security releases for Asterisk 16, 17 and 18, and Certified Asterisk 16. co. GitHub Gist: instantly share code, notes, and snippets. "gsm,h264". I can't think of any free softphones that supports g. If you are on an x86 server, you can enable opus in make menuselect, or download it from the github project, otherwise take the opus codec out of the allow= section of the endpoint. 4. Two Digium phones were involved, the phones themselves were both configured to have ulaw,alaw,g722 codec preference order. I do some simple configuration on Asterisk Sever: Add four accout for two Pjsip phone and my SjPhones. Pastebin is a website where you can store text online for a set period of time. Scenario: VPS, No nat, minimal Debian 8(Jessie), Trunk to Telecube, One phone behind nat, no voicemail or other features. hogetara. Technically, yes. 0: Pjsip: Unnecessary 603 Decline Because Of Wrong Codec Decision Looking For The Carrier That Owns A Particular DID >> 2 thoughts on - Pjsip Insecure=port,invite Joshua Colp says: Apr 24, 2020 codec g711ulaw. 28 replies pjsip list ciphers -- List available OpenSSL cipher names: pjsip list contacts -- List PJSIP Contacts: pjsip list endpoints -- List PJSIP Endpoints: pjsip list identifies -- List PJSIP Identifies: pjsip list registrations -- List PJSIP Registrations: pjsip list transports -- List PJSIP Transports Any changes to bindings and transports in Settings, Asterisk SIP Settings requires an Asterisk restart after the Apply Config. Default is 16000. 7. txt and codec_tests. 4 net =0 15. The Asterisk is version 11 LTS and it is a vinilla installation. so Revision: 406848 Reporter: danjenkins Coders: kharwell ASTERISK-23489: Vulnerability in res_pjsip_pubsub Asterisk 13. e. DEPRECATED: Asterisk 15. 16. 2-rwxr-xr-x 1 root root 112292 May 10 02:06 libpjlib-util. [asterisk-dev] Wish: adding intelligent codec negotiation to asterisk / pjsip. Asterisk Rest Interface) + WS (for events). 51 is because the asterisk limits do 50 calls to the func “SipAddHeader” and this way it wont interfere Example: Improve codec negotiation between Asterisk servers Note: This example has little practical use because the codec (audionativeformat) cannot be reliably determined before the call has been answered, and after that setting the SIP Asterisk comes with two different SIP modules, a standard SIP module and the PJSIP module. 264 VideoToolbox codec; PJSIP version 2. Someone help me !!!, I have my s300 working for almost a year, from one day to the next I stop making calls and receiving through trunk E1. Codec negotiation in Asterisk has been one of its deepest darkest secrets. 2 (since it is stable and working for us), but I have also tested with 1. 5 * 6 * Mark Michelson <mmichelson@digium. There are some hints provided internally for outgoing legs but the result is still ultimately independent. pjsip. example. 1 is released with support for BB10, SILK, and OpenCore AMR-WB » PJNATH adds STUN, TURN, ICE to Asterisk for WebRTC support - During this time while asterisk was sending audio in G722 format, the T46g phone was sending back in G711 format; - After the initial announcement played, Asterisk went back to sending audio in codec G711 and I then could here the echo test. . [YMCS/YDMP Free Trial Program]Yealink would like to offer Free Trial Program of Yealink device management service for our current eligible customers. 0 to ports - Update g72x module to 1. Asterisk Codecs. [ASTERISK-29109] – res_pjsip_session: Asterisk 18 does not progress calls due to codec negotiation after upgrading from Asterisk 16 (Reported by Ross Beer) [ASTERISK-28430] – res_rtp_asterisk. Which will bind UDP port 5060 (in default configuration) to ip address of a device with Asterisk installed. It has an old Intel atom processor running on CentOS 6. The Peer’s phone has selected only uLaw and speed in this case. Asterisk is an open source communications project enabling administrators to create telephony applications for IP-based private branch exchanges (PBXs), voice over IP (VoIP) gateways and conference servers. com t=0 0 m=audio 49170 RTP/AVP 0 8 97 a=rtpmap:0 Hi, I am currently run FreePBX12/Asterisk13. Yes. 7. Codec ENUM(E. so Dialplan mutexes 0 Running core func_logic. 0, and Certified Asterisk through 13. ASTERISK-26556 - manager: AMI [prev in list] [next in list] [prev in thread] [next in thread] List: asterisk-dev Subject: [asterisk-dev] RFC: pjsip show endpoints output format From: ; PJSIP Configuration Samples and Quick Reference 2; 3; This file has several very basic configuration examples, to serve as a quick 4; reference to jog your memory when you need to write up a new configuration. Loading… Dashboards Line 1 /* 2 * Asterisk -- An open source telephony toolkit. endpoint. Settings-->Asterisk SIP Settings-->General SIP Settings (tab): * Set the eight CODECs from the ADP active (check marks) and disabled the rest * Set the priority of the CODECs to match the common priority in all three Grandstream devices. My cluster is E. in general you need to have a codec enabled on the system able to negotiate with the incoming invite. While an SDP negotiation may result in a codec using a different payload number these desired ones are still stored internally. Colp [asterisk-users] Realtime PJSIP max_streams' issues Ahmed Chohan. so An SDP negotiation vulnerability in PJSIP allows a remote server to potentially crash Asterisk by sending specific SIP responses that cause an SDP negotiation failure. As mentioned above, because the audio path includes Asterisk, an extra negotiation occurs. A memory leak occurs when an Asterisk pjsip session object is created and that call gets rejected before the session itself is fully established. 10. 7; 8 An SDP negotiation vulnerability in PJSIP allows a remote server to potentially crash Asterisk by sending specific SIP responses that cause an SDP negotiation failure. txt was used to make the calls in full. 7. Media negotiation state information has been changed to b ソニーのロケーションフリーテレビについて、「お風呂で使いたいのに」と書いたのですが、なんとカタログによると「lf-x1専用お風呂ジャケット」なるアクセサリーがあるのですね。 Option reference for all PJSIP modules. 16. 7. However, in Asterisk the method that handles this decrements the reference. 4. Asterisk is a PBX server program to manage phones. Like chan_pjsip in 12 and the XMPP Jingle drivers. This is a general package update to the CURRENT release repository based upon TrueOS 19. transcoding in Asterisk 1. The res_pjsip module handles configuration, so we'll mostly speak in terms of configuring res_pjsip. 13 before 13. I have also noticed that crashes happens even though no calls are made, so I don't think it has anything to do with the dialplan. An SDP negotiation vulnerability in PJSIP allows a remote server to potentially crash Asterisk by sending specific SIP responses that cause an SDP negotiation failure. 1 with PJSIP and incoming calls from 3 different SIP providers that were working before and only use alaw and ulaw show the following error: [2018-10-30 12:14:34] WARNING[9942][C-0000002b]: chan_pjsip. Introduction. Hello, I set up asterisk 16 and jigasi using a pjsip channel following the example in Working configuration for Jitsi/Jigasi with Asterisk/SIP. A limit has been set on Outbound INVITEs so that, once reached, Asterisk will stop sending INVITEs and the transaction will terminate. x will reach EOL on 2019-10-03. In practice it’s a bit iffy – specifically because some DSPs in devices won’t allow it – they require a single codec be in use for each direction. It’s a lot so please read carefully. The order of the codec specifies the priority. 13 before 13. 4 that allows SIP_CODEC to contain a list of codecs , e. so Audio Effects Dialplan Functions 0 Running extended func_pjsip_aor. In file sip. Other notes: - If on Asterisk I make G722 the first option than everything works fine. The channel driver itself being chan_pjsip which depends on res_pjsip and its many associated modules. 4. c : Memory leak if endpoint not found Re: [asterisk-users] Asterisk not using common codec between (SIP) endpoints Joshua C. 3_3 Version of this port present on the latest quarterly branch. 0 Running core func_lock. you can do it, for that you should use only pjsip core library and remove other libraries from the pjsip such as pjmedia,codec,nath etc. 7; 8 - Add asterisk 16. conf. de:5072 Software: kamailio 4. 13-cert7. 0. And install two SjPhones,One on my PC,the other one on another PC. The reason is that Asterisk 13 supports two forms of SIP, the older type we are all used to and a newer one called PJSIP. 0: Pjsip: Unnecessary 603 Decline Because Of Wrong Codec Decision Looking For The Carrier That Owns A Particular DID >> 2 thoughts on - Pjsip Insecure=port,invite Joshua Colp says Hi, We are trying to setup PJSIP with our SIP trunk. 0, and Certified Asterisk through 13. org] On Behalf Of David Clark Sent: Saturday, August 15, 2009 7:18 PM To: pjsip list; 'pjsip list' Subject: Re: [pjsip] In-band DTMF detection Ok I have not done inband but I have done FFT and they are similar in terms of approach. representation of codec negotiation preferences, which may come from a pjsip endpoint for example, and populates a ast_stream_codec_negotiation_prefs structure. so HTTP WebSocket Support 3 Running extended res_pjsip > Hello! > > On 20. c: FRACK!, Failed assertion errno != EBADF (Reported by under) [ASTERISK-29108] – resource_endpoints. Asterix PBX install sudo apt-get install alsaplayer-alsa python2. I should also say that this occurs in codec_negotiation = strict as well as codec_negotiation = loose So asterisk knows that it is going to dial a peer that supports only g729 when it gets an invite from a peer that supports both ulaw and g729. Sangoma Connect is only supported for User Management users whose primary linked extension is of type PJSIP. I have been informed by Grandstream that enough people need to request the addition of the Opus audio codec in order to add it. ; * ASTERISK-25702 - PjSip realtime DB and Cache Errors since upgrade to asterisk-13. 2. - Add asterisk 16. Other RTP-channels Most of this work is generic in Asterisk. x and 15. started 2017-01-30 10:13:28 UTC. So configuring my I am using asterisk 1. 4. Top Features of Opus Even with pjsip video guide, Aug 15, 2019 · Project Trident 19. Google uses Opus as a standard in Chrome. demand video resolution is 1680x1050, 1600x1200, 1920x1080. 5; It is not intended to teach PJSIP configuration or serve as an exhaustive 6; reference of options and potential scenarios. 4. so Logical dialplan functions 0 Running core func_math. conf: [general] context=default [7001] * ASTERISK-26691 - Remember SDP negotiation on SIP_CODEC_INBOUND. 8 is released with WebRTC interopability - RTP/SAVPF - SSRC and OPUS param on the fly Después de la instalación de Asterisk 12, ya podemos realizar la primera prueba de llamadas entre extensiones configuradas en PJSIP. jp ; Tell Asterisk which context to use when this peer is dialing: directmedia=no ; Asterisk will relay media for this peer Powered by a free Atlassian JIRA open source license for Asterisk. 18. I see the Video Codecs being forwarded by my soft client to the server and I have H264 & VP8 enabled under the Asterisk SIP settings configuration, as well as in the extension allowed field; however, the GUI settings for the PJSIP trunk group ( CODECS ) are filtering the video CODECS from the INVITE request. 5; It is not intended to teach PJSIP configuration or serve as an exhaustive 6; reference of options and potential scenarios. Codec negotiation in Asterisk has been one of its deepest darkest secrets. 11. What is Opus? Opus is a recently-developed audio codec, many people call it the “codec of the future”. Codec choices – the benefits of G. I'm using pjsip on Asterisk. you can check this issue by running following command, I don't see any error but you can try following command to check RTP communication ASTERISK-25455: Deadlock of PJSIP realtime over res_config_pgsql Reported by: mdu113 [9f4892ece4] mdu113 -- res_config_pgsql. I am now running mine without any codecs loaded at all. 16. so Cryptographic Digital Signatures 1 Running core 1 modules loaded # asterisk -rx "module show like websocket" Module Description Use Count Status Support Level res_http_websocket. 0-rc2 (Reported by Nic Colledge) * ASTERISK-25730 - build: make uninstall after make distclean tries to remove root (Reported by George Joseph) Media quality, 0-10, according to this table: 5-10: resampling use large filter, 3-4: resampling use small filter, 1-2: resampling use linear. When Asterisk sends a reinvite negotiating T38 faxing, it's possible a. com> - update to 18. Given a scenario where an outgoing call is placed from Asterisk to a remote SIP server it is possible for a crash to occur. 13; How to do Multiple outbound registration in Astersik 13. 729, but I zoiper supports g. 2, 17. When set to "yes" the codec in use for sending will be allowed to differ from that of the received one. This page and its sub-pages are intended to help an administrator configure the new SIP resources and channel driver included with Asterisk 12. Because in pjsip. 38 re-invite. When Asterisk sends a reinvite negotiating T38 faxing, it's possible a. Do you need a new Asterisk feature? If you need a new Asterisk feature or want to get general Asterisk or Kamailio PJSIP also provides three main components of real-time multimedia application, i. com> 7 * 8 * See http PJSIP is a free and open source multimedia communication library written in C language implementing standard based protocols such as SIP, SDP, RTP, STUN, TURN, and ICE. Como PJSIP y el canal SIP, por defecto, escuchan en el puerto 5060, tenemos dos opciones: desactivar el modulo chan_sip utilizar un puerto distinto al 5060 para PJSIP En este caso se ha # adduser asterisk -c "Asterisk User" # passwd asterisk # usermod -aG wheel asterisk # su asterisk Next, install PJSIP, is a free open source multimedia communication library that implements standard based protocols such as SIP,SDP,RTP,STUN,TURN, and ICE. Aqui você irá encontrar muito conteúdo, tutorias, how-to, manuais, dicas e reviews de vários produtos e fabricantes. In Mediabroker, This is done in the calls. I looked at Asterisk again after about 10 years since the last time. x through 16. A pointer dereference in chan_sip while handling SDP negotiation allows an attacker to crash Asterisk when handling an SDP answer to an outgoing T. 729 codec for Asterisk PBX based on audio/bcg729. Using the offer / answer model it would look like this: Peer -> Invite (SDP:ulaw,g729) -> Asterisk Peer <- 100 Trying (w/ SDP -- g729 only) <- Asterisk Peer -> 200 OK (w/ SDP g729) -> Asterisk This page and its sub-pages are intended to help an administrator configure the new SIP resources and channel driver included with Asterisk 12. Warning: Asterisk has only basic WebRTC support and doesn't handle corner cases such as streaming over HTTP port 80 (which is needed for most corporate networks where UDP is blocked) and also it doesn't have a built-in TURN server (a separate TURN server needs to be installed). x support; PJSIP version 2. 711 alaw (as used in Europe); G. An issue was discovered in Asterisk Open Source through 13. 23. The additional advanced codec negotiation options have also been removed from the sample configuration and marked as reserved for future functionality in XML documentation. 08 Now Available. Also pjsua-lib will now reports to application via a callback when ICE negotiation has failed (ticket #370). E-Learning Echo Test with media simultaneous calls 3338 3076 2820 2700 2700 0 500 1000 1500 2000 2500 3000 3500 4000 Asterisk 11 Asterisk 13 Asterisk 15 chan_sip chan_pjsip 45. La configuración es bastante distinta a la que estamos acostumbrados. x through 15. asterisk pjsip codec negotiation