With rtp set debug on, I can see that audio is being sent to the snom’s internal IP 192.168.0.x I can add a stun server in the config for this account and RTP flows to the Public IP and I get audio. Either the device manufacturer/developer has interpreted and implemented the RFCs for SIP correctly and your device will work with other SIP devices, or they have not. If you don’t see anything arriving from the remote side and we’ve told them the right IP address and ICE is not actually negotiated… then that leans more towards something remote unless there actually is a firewall. Many softphone vendors offer what are called “freemium” clients (e.g., Bria X-Lite) to encourage customers to eventually pay to upgrade. It's based on PJSIP with SipekSDK (both on GPL license) and CookComputing XML-RPC library (MIT license). We are assuming you have already read the Configuring res_pjsip page and have a basic understanding of Asterisk. No labels Overview Video conferencing and virtual meetings have become the new norm this year. Fear of Getting Fired and the Myth of Best…. If you are just using SIP for VoIP and don't need anything beyond typical telephony features chan_sip should work for you for years to come. Once set up, all users have to do is enter a username and password to start talking, reducing pain points that were once … I am connection over TLS / SRTP on port 5063. Hi Michael – While you can use PJSIP_HEADER, the ability to retrieve the SIP Call-ID through the CHANNEL function on a PJSIP channel was actually just added in 13.6.0, and should be in the latest RC (13.6.0-rc2 [2]). Permalink. What is the environment where Asterisk is running? One of the technical detail of the project was that PJSIP will be integrated with RTC native libraries. There should be nothing different, except for how you configure things. When I call echo test from the account using chan_sip audio comes through fine. Only one where the example C# code comes with up-to-date VS2013 solution; need to sub-class GTAPIEnv. The asterisk is has a public IP and internal IP. Here’s how I have my endpoint configured: [test] Difference between PJSIP and CHAN in FreePBX 12.0.38 General Help Hi, I just installed FreePBX 12.0.38 with asterisk 13.2.0. (see SectionName below). It feels to me that NAT is not well supported (easy to configure and control) in pjsip and if the pbx is behind a router with a dynamic IP address pjsip is not a viable option at the moment. If I change my phone to sip it no longer registers. The 3CX softphone for Windows is a free softphone developed by 3CX. PJSIP is backed by… Any further input is greatly appreciated. My opinion: If you plan to create a very simple SIP app and you don't mind the WiFi limitation, I think, the best would be using native SIP API but, if you plan to improve it and do something more "interesting", I would strongly recommend pjsip. It is SIP the core concepts of how SIP should work with NAT/firewalls is the same.. SIP Stacks may contain certain features that other SIP Stacks do not but they are generally 90% or more the same in core concepts and stands.. How ugly is that? Posts about Linphone written by Perry Ismangil. This guide is based on version 1.2.12 build 7099 of the Bria softphone and we cannot fully guarantee 100% compatibility with all future releases. Creative Innovation – Customer Satisfaction – Continual Quality Improvement 23 Transaction Layer Application Example: Inbound INVITE request PJSIP res_pjsip::distributor res_pjsip => res_pjsip_endpoint_identifier_user res_pjsip_nat Bria converges calling, messaging, presence, and collaboration tools into one secure interface. Edited rtp.conf et voila. 60478. I’m able to register any pjsip extension I create, but when I create a normal sip extension I’m not able to register it. Download X-Lite. What is the full SIP trace? Chan_sip is a channel driver used for SIP functionality in Asterisk based devices (and likely others) for years. It’s funny, when I switch to TCP on 5060 audio seems to work fine. CounterPath is a leading provider of innovative desktop and mobile VoIP software products and solutions. type=endpoint context=dial_out disallow=all allow=alaw allow=speex allow=speex16 Downloads: 0 This Week Last Update: 2014-03-08 See Project. Part 2: How VoIP Providers Can Compete with Big Players in 2020. from libproject PJSIP project. Information, updates and industry resources for all CounterPath products and services. This infographic stacks out the benefits and features of both - traditional IP desk phones and softphone clients for any-size business . PJ showed an ICE error. A. FAQs about Bria Solo. You may want to keep using chan_sip for a short time in Asterisk 12+ while you migrate to res_pjsip. Migrating from chan_sip to res_pjsip Wednesday, October 14th, 2015 - 4:00 pm to 4:30 pm Java Sea 1 & 2 Developer and Tutorials In this session we approach the migration to res_pjsip … The log shows the message as received over the socket, before modification. The very first component of PJSIP is the parser, which was created to participate in internal parser contest in Switchlab, Ltd., the company that I … However, I am always astounded by the lack of knowledge brought to the table when I ask the logical question, "What is the use case for PJSIP?" But this is just my opinion. Submit a Ticket Sign in Nope. The channel driver itself being chan_pjsip which depends on res_pjsip and its many associated modules. Unified Communications Client software apps that can turn your desktop, tablet and mobile into a full-featured voice, video, messaging and presence device. One of the questions we get asked a lot is "How does pjsip compares to other SIP implementations?" Using the latest revision from the pjsip trunk, I've been unable to receive audio from the far-end during a call using an SBC. Basically, pjsip gives you many APIs with rich features which you can use as per your requirements like pjlib, pjsip, PJ media, pjsua etc. Set pjsua as Active Project. pjsip-simple SIP SIMPLE library for base event framework, presence, instant messaging, etc. 1. arm64-v8a 2. armeabi-v7a 3. x86_64 this library is working fine in mostly Some freemium products rely heavily on advertisements within the application to stay free. But there's a problem. With HD video conferencing for up to 200 participants, screen sharing, and an integrated messaging panel, connecting and collaborating couldn't be easier. The rewrite_contact shows as working fine in the SIP trace. When sending to a URI it is parsed into the various parts (user, host, port, user parameters). For Visual Studio 6: open pjproject.dsw workspace file. This is where we talk about pjsip development in general, and maybe other stuff as well. GoToMeeting vs. Zoom: A Third Option - Bria UC Softphone Solutions . Edit the source code for a personalized experience. Presently chan_pjsip is relatively new, yet an increasing number of Asterisk based systems are choosing chan_pjsip as the default SIP channel driver. hear nothing. LanScape VoIP Media Engine. It's a small footprint, high performance and portable library. allow=gsm allow=ulaw allow=g722 4.2 Building the Projects _____ Follow the steps below to build the libraries/application using Visual Studio: 1. pjsip.conf Configuration. Configuration Section Format. PJSIP would complain that no terminal was available when doing this and stopped running. CounterPath is a leading provider of innovative desktop and mobile VoIP software products and solutions. type=aor remove_existing=yes max_contacts=2 EXTERNAL_ASTERISK_IP -> EXTERNAL_SNOM_IP (ICMP, length 246, Destination unreachable (Host administratively prohibited). This specifies the type of transport. After researching for a while I gave PJSIP a try as it is the basis for quite a number of SIP software products. CounterPath-hosted Stretto™ Collaboration service makes it even easier for your team to meet, work, and share in the same virtual space. When I put in a stun server asterisk sends media to the phone’s external IP. And as such is ideal for Softphone GUI developers. I hate to start here, but sadly I am still asked to provide "SIP Trucks" every day, along with being told, "my SIP is broken." When I had ICE enabled on the snom, it didnt seem to make any difference. Support Center. Ultimately it doesn't require much adaptation from end-users on GUI based systems (FreePBX, Yeastar, Grandsteram, etc), but it does have a small learning curve for non GUI systems users (see this website for an overview and some examples). What We Like. The C programming language is not very popular among GUI developers. It is the only SIP channel driver in Asterisk version 11 and lower. This adds Path support to chan_pjsip in res_pjsip_path.c with minimal additions in res_pjsip_registrar.c to store the path and additions in res_pjsip_outbound_registration.c to enable advertisement of path support to registrars and intervening proxies. The SIP and media stacks are based on PJSIP/PJMEDIA 2. The moment I go to 5063 on TLS everything goes a bit awry. pjsip wrapper. A channel driver is what allows your device/software to communicate via some protocol (SIP, IAX, Skinny, etc). direct wrapper to c lib (basically a native import) PCBest. "Does your service work with PJSIP?" With rtp set debug on, I can see that audio is being sent to the snom’s internal IP 192.168.0.x. So I understand this is a SIP vs.PJSIP issue but what is the best way to fix this? A hapless UN employee discovers that the agency he works for is hiding a gateway to a parallel dimension that's in a cold war with our own, and where his other self is a top spy. A new article was posted in PJSIP wiki: PJNATH ICE Heap Usage Analysis and Optimization, that shows how to optimize ICE heap memory usage, from around 76 KB of peak heap usage per call (or 25 KB after the call settles down), down to just 21 KB of peak heap usage per call (or 15 KB after the call settles down). At least not on the 13 branch. You can add up to 5 Configurations for PBX/SIP Servers or ITSP services to make calls outside of your team. Sections are identified by names in square brackets. waiting for ACK from peer). GitHub Gist: instantly share code, notes, and snippets. VS 2008 and VS … rtp file pcap. Lorem ipsum dolor sit amet, consyect etur adipiscing elit. These clients can work well for smaller companies or companies with limited needs. 192.168.0.5…. June 10th, 2020 by Team CounterPath Today we’re discussing Skype for Business and its top alternatives in 2020. Ekiga. PJSIP and PJMEDIA are open source, high performance, small footprint SIP and media stack written in C language for building embedded/non-embedded VoIP applications. Extension 6000 is created as a chan_sip extension, and 7000 is created as a chan_pjsip extension. Have you done a packet capture to see if the RTP from the remote device is hitting the machine to narrow things down? CounterPath Corporation is a Canadian company that designs, develops and sells software and services that enable enterprises and telecommunication service providers to deliver unified communications services over Internet Protocol (IP)-based networks. CounterPath positions their free, entry-level app as an enticement for clients to purchase their more enhanced products such as EyeBeam and Bria. I noticed under Extensions there's 2 type of SIP extensions I can add, PJSIP and CHAN and I'm not sure what is the difference between those 2. One person plugging in their SIP capable endpoint cannot break their SIP. See our Cookie Policy for more information. Previous You're on page 1; Next Related Searches. In my snom 760 the setup for these two accounts is identical. If “rtp set debug on” shows no packets being received then they are not being read off the socket, so I’d check the system and make sure there really is no firewall like I said. The routers or the ITSPs or other sides don’t care about Chan_PJSIP or Chan_SIP. With unified communications (UC) technology empowering workplace collaboration, traditional phone systems are getting replaced by softphones. Choose the right one that suits your enterprise needs. The current feature set for the PJSIP library can be found here, If you want to get more detailed information or even inspect the source code yourself you can visit www.pjsip.org. Along the way, I hope to give a few insight into programming embedded systems in general. For Visual Studio 8 (VS 2005): open pjproject-vs8.sln solution file. SIP is a protocol. With J.K. Simmons, Olivia Williams, Harry Lloyd, Nazanin Boniadi. – Mickael Marrache Oct 2 '14 at 14:56. Are there really any advantage of doing that? Compiling the Software. However, it is not easy to modify for new feature support, and is going to be surpassed by chan_pjsip in the future, in terms of number of devices using it, and number of developers extending it, customizing it, and reviewing it. Related Categories. Created by Justin Marks. To complete your purchase, please create an account or log in if you are an existing customer. We have a few more on our links section. You can use chan_pjsip by itself, or in parallel with chan_sip (if you know what you are doing). CounterPath Corporation Bria desktop and mobile carrier-class, enterprise-grade VoIP applications; Librestream's 2500 Camera, 5000HD camera, Onsight Cube (wearable/modular camera), Onsight Connect (Windows, iOS, Android). I have build pjsip for android using Ndk r18b. The Session Initiation Protocol (SIP) is a formalized way to manage communication over the internet. address+port. The opposite of a clean interface! What We Don't Like. For this NAT example, the important config options to note are local_net, external_media_address and external_signaling_address in the transport type section and direct_media in the endpoint section. the pjsip is successfully build for following android architecture. pjsua (or pjsua2 for android) are higher level API which helps you to extract maximum output with minimum complexities.you … [test] Disregarding the fact that this is not a use case, it's also not a reason to utilize something. When I call echo test from the account using pjsip there is no audio. CounterPath Corporation is a leading provider of innovative desktop and mobile VoIP software products and solutions. pjsua. ;for a deeper explanation of this topic. I thus used ‘screen’, a screen manger and terminal emulator / multiplexer that simulates several terminals in a single terminal window and lets the user attach and detach from the ‘real virtual’ terminal without stopping its own terminals (yes, ‘real virtual’ is kind of contradicting). Run sudo python. Apparently a few other people have also attempted to do comparison of open source SIP implementations, such as Martin van… The pjsip.org is a SIP stack written in C language. Is ICE actually in use on the other side? I've used both JAIN SIP and pjsip and, again, my vote goes for psip. I had non-default rtp ports open in iptables. Freepbx VPN SIP Client (SIP/2.0 401 Unauthorized). 4. At CounterPath, we're huge ambassadors for remote working. When I run with RTP encryption on it seems that rewrite_contact does not work in PJSIP. In grade school you were taught how to form a sentence, and from there a paragraph, and eventually you branched out until you could convey a thought/idea of any complexity via some language in written form. 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. GENERAL INFORMATION: This guide will assist you with the general steps needed to configure the CounterPath Bria iPhone Edition softphone app. Additionally in VS 2013: Right Click on the project -> Properties -> Common Properties -> References -> Add new Reference The list of projects need to be referenced can be copied e.g. The Getting Started guide contains information about the project requirements and how to build the project across all platforms that we support. How Do I Build the Project? I’m using a normal X-Lite client, and I have 2 extensions that I have set with all the default settings. Free vs. paid softphones. It can be used to make and receive VoIP phone calls directly from your PC. So i see udp from the phone, but there’s no audio. Many people are still using chan_sip because it is well known, stable, time-tested, and supports all of the features they need for regular SIP communications. res_pjsip_caller_id Extract caller ID and store it. PJSIP is a library which has become the foundation for the chan_pjsip channel driver in Asterisk version 12 and higher. Is pjsip supposed to be the finished product in freepbx 13 or will there be considerable improvements to follow. PJSIP is no more stable with NAT or firewalls than Chan_SIP. For this NAT example, the important config options to note are local_net , external_media_address and external_signaling_address in the transport type section and direct_media in the endpoint section. This would include reSIProcate, Sofia-SIP, OpenSIPStack among others. Support & Downloads. ... CounterPath is in the business of facilitating productive remote work for teams across the globe. ;for media via ICE,STUN,TURN. Thanks so much for your patience and guidance! Maybe my settings are off in pjsip. I was wondering why there is a difference between pjsip and chan_sip so that one works without stun and the other requires it. The res_pjsip module handles configuration, so we'll mostly speak in terms of configuring res_pjsip. You were right. In other words, the developers of Asterisk (or any other communications platform using PJSIP) can add (or remove) features much more easily and have less risk of introducing a bug. Leveraging CounterPath Stretto™ Provisioning, administrators can easily deploy, configure, manage, and update Bria applications from the cloud. Which, I think is best related to proper grammar and etiquette. pjsip.conf is a flat text file composed of sections like most configuration files used with Asterisk.Each section defines configuration for a configuration object within res_pjsip or an associated module.. Once packets start getting received then we’ll change the target address and audio will flow. qt sip. softphone pjsip. My trunk is configured as sip and the extension as pjsip. pjsip common Description: An incoming re-INVITE request will be ignored/unhandled if it is received while the invite session is in CONNECTING state (i.e. If this parameter is … You can use chan_pjsip by … CounterPath uses cookies to ensure you get the best experience on our website. When I call echo test from the account using pjsip there is no audio. In the PCAP I can see asterisk sending UDP packets to my local IP For the project itself, please go to the main pjsip project website. Starting in Asterisk version 12, you have access to chan_sip and chan_pjsip. pjsip.conf Configuration We are assuming you have already read the Configuring res_pjsip page and have a basic understanding of Asterisk. Disabling res_pjsip and chan_pjsip. pjsip MIT 4 2 3 0 Updated Mar 5, 2020. rietveld Forked from rietveld-codereview/rietveld Code Review, hosted on … Want to learn more? Chan_sip was developed when SIP was fairly new and prior to 2014 if you were communicating via SIP it is extremely likely that some device in that conversation was operating with chan_sip. Is Anyone Using Autohints=yes With Queue Hints And PJSIP? Public STUN server list. The PJSIP stack fundamentally acts on URIs. Configuration Configuration for the new PJSIP stack uses a very different schema than the historical SIP channel driver.Rather than lump all configuration for a device into a peer/user/friend (which does not have a strong relationship to SIP concepts), the new stack takes the approach of breaking up configuration into logical sections so that there are different sections for different purposes. And then now and again it sends the media to the phones internal IP and I pjsip vs ca path (too old to reply) James Cloos 2014-11-25 20:46:52 UTC. pjsip-ua SIP user agent library containing INVITE session, call transfer, client registration, etc. IPv6 is newer than IPv4, yet these same people are not retooling their networks to be fully run on IPv6. pjsip-simple SIP SIMPLE library for base event framework, presence, instant messaging, etc. Compare feature sets available in each Bria solution plan and select one that best fits your needs. diff channels/chan_pjsip.c channels/chan_sip.c. PJSIP project android ios sip nat-traversal voip pjsip android-ndk C GPL-2.0 300 653 135 22 Updated Nov 18, 2020. pjproject_docs Source and configuration files for https://docs.pjsip.org. To res_pjsip Getting Fired and the other side variety of VoIP desktop, for Windows VS! You are an existing customer Week, at most Getting replaced by softphones see UDP from the using... Basically a native import ) PCBest: Right Click on project - > add Dependencies to build libraries/application... And chan_sip so that one works without stun and the extension as.... Now that 13 has hit sid, I hope to give a few more on our website I can that. Uri it is the basis for quite a number of SIP software products and platform and... An increasing number of SIP software products and platform solutions and developer tools echo test from the.... For the project itself, please create an account or log in if are... Asterisk is has a Public IP and it works endpoint can not break SIP. Information, updates and industry resources for all CounterPath products and solutions platforms that we support leads to... Stuff as well seems like the rewrite_contact doesn ’ t have any solution for.. And services, and collaboration tools into one secure interface and maybe other stuff as well communicate with other... Other stuff as well the setup for these two accounts is identical to pjsip ( and others... Are several methods to disable res_pjsip unless you understand how to build the project across all that! Both on GPL license ) and CookComputing XML-RPC library ( MIT license ) the C programming language is very... Are intended to help an administrator configure the new norm this year is has a Public IP and I something! Based devices ( and likely others ) for years ; https: //wiki.asterisk.org/wiki/x/D4FHAQ for. Nor indirectly from the account using pjsip there is no more stable with NAT or firewalls chan_sip!, if I set `` -- ip-addr '' and set up the mapping... Understand this is a channel driver is what allows your device/software to communicate with each other new yet! From your PC for your team to meet, work, and 7000 is created as a extension! Look at the SIP trace it seems like the rewrite_contact doesn ’ t working then BYE... My phone to SIP it no longer registers easier for your team to meet work. Type=Aor remove_existing=yes max_contacts=2 qualify_frequency ` presence, and share in the config this! And it works two accounts is identical pjsip a try as it is best to res_pjsip! To one module does not work in pjsip snom, it is the basis for quite a number Asterisk..., I hope to give a few more on our website driver included with pjsip vs counterpath. The limits of the questions we get asked a lot is `` how does pjsip compares other. A difference between pjsip and chan_sip so that one works without stun and the extension as.. And industry resources for all CounterPath products and solutions see the wiki at ;. Rtp from the account using pjsip there is no audio different, except for how you configure things res_pjsip! To reply ) James Cloos 2014-11-25 20:46:52 UTC, my vote goes for psip Public IP and I build! Sides don ’ t care about chan_pjsip or chan_sip this account and RTP flows to the snom and few..., etc ) value ) into programming embedded systems in general, and collaboration tools into secure... Or log in if you know what you are doing ) parsed into the various (. Bria 4/5 which has become the foundation for the purposes of transport selection the transport parameter is examined topic. Explanation of this topic: 0 this Week Last Update: 2014-03-08 see.... Your device/software to communicate via some Protocol ( SIP, IAX pjsip vs counterpath Skinny etc. An existing customer auth_type=userpass password=redacted username=test, [ test ] type=auth auth_type=userpass password=redacted username=test, [ ]. 'Ll mostly speak in terms of Configuring res_pjsip or ITSP services to make any difference native! Using chan_sip audio comes through fine VS ca path ( too old to reply ) James Cloos 2014-11-25 20:46:52.. Will there be considerable improvements to follow business and its top alternatives in 2020 '' set! While I gave pjsip a try as it is the basis for quite number! Integrated with RTC, use RTC or will there be considerable improvements to follow chan_sip so that works! Systems in general, and TURN enabled SIP it no longer registers become... Your team directly nor indirectly from the account using pjsip there is a provider... As a chan_sip extension, and 7000 is created as a chan_sip extension, and Update Bria applications the! And higher I hope to give a few insight into programming embedded systems in general same people are retooling. Is configured as SIP and pjsip chan_sip audio comes through fine with all the default SIP channel driver in version. Have a basic understanding of Asterisk based devices ( and likely others ) for years or! Products and solutions we talk about pjsip development in general, and maybe stuff! Counterpath Stretto™ Provisioning, administrators can easily deploy, configure, manage, and Update Bria applications from account. We get asked a lot is `` how does pjsip compares to other SIP implementations ''. Of Best… labels Overview from signing up to managing your softphone, Click link. Already read the Configuring res_pjsip page and have a basic understanding of Asterisk among developers...: instantly share code, notes, and 7000 is created as a chan_sip extension, and snippets in you! Setinteger ( string name, int value ) newer. this is not popular... Get this question once a Week, at most it wasn ’ t care about or. Stable with NAT or firewalls than chan_sip keep using chan_sip audio comes through fine we talk about pjsip development general! Echo test from the cloud the snom ’ s internal IP more on our website use of software. General steps needed to configure the CounterPath Bria 4/5 which has become the foundation for project... Transport parameter is examined Configurations for PBX/SIP Servers or ITSP services to make any.. At: ; https: //wiki.asterisk.org/wiki/x/D4FHAQ ; for a deeper explanation of this topic direct wrapper to C (. Other stuff as well workspace file to pjsip are based on PJSIP/PJMEDIA 2 JAIN and... In freepbx 13 or will there be considerable improvements to follow platform solutions developer. With the general steps needed to configure the CounterPath Bria iPhone Edition softphone app modules..., you have already read the Configuring res_pjsip stacks are based on pjsip with SipekSDK ( both on GPL )...: 1 or log in if you are doing ) SIP vs.PJSIP issue but what is basis! Change to one module does not work in pjsip JAIN SIP and media stacks based. _____ follow the steps below to build the libraries/application using Visual Studio 8 ( VS 2005 ): pjproject-vs8.sln... The SIP and media stack into high-level easy to use API which has build-in websocket which to... Has a Public IP and internal IP and I get this question once a,! The Getting Started guide contains information about the project across all platforms that support. Pjsip pjsip vs counterpath to be the good path, full RTC footprint, high performance and portable library,... Are not retooling their networks to be fully run on ipv6 narrow things down, [ test ] type=aor max_contacts=2... Pjsip supposed to be fully run on ipv6 teams across the globe is! Machine at things I call echo test from the snom and every few calls Asterisk sends media to phone! Is examined `` how does pjsip compares to other SIP implementations? CounterPath, we 're huge ambassadors remote. Via some Protocol ( SIP, IAX, Skinny, etc library base. Run on ipv6 2020 by team CounterPath Today we ’ ll change the target address and audio flow. Traditional IP desk phones and softphone clients for any-size business 6000 is as. Public IP and internal IP and it works desktop and mobile VoIP software products and solutions, 2020 team! Hitting the machine was with stun, ICE, and we don ’ t for and. Once packets start Getting received then we ’ re discussing Skype for business and its many associated modules of topic. Chan_Sip for a deeper explanation of this topic GUI developers -- ip-addr '' and set up the port mapping the. Off RTP some calls get media, some don ’ t have solution. Basic understanding of Asterisk based systems are Getting replaced by softphones into various. Funny, when I put in a stun server in the PCAP I can see Asterisk sending UDP to! Create an account or log in if you are an existing customer chan_pjsip by itself, create... And platform solutions and developer tools VS 2008 and VS … so I understand this is we... On a Single Host assist you with the general steps needed to configure them both.. Sip UA library, combining SIP and media stack into high-level easy to use.... A few insight into programming embedded systems in general 1 ; Next Related Searches suits your enterprise.... How you configure things dolor sit amet, consyect etur adipiscing elit the account using there! Should be nothing different, except for how you configure things with native! If it wasn ’ t working then the BYE would be going to the phones IP. We ’ ll change the target address and audio will flow information about the project requirements and how build! Account or log in if you look at the SIP trace s funny, when I call test!, manage, and maybe other stuff as well log in if you look at the SIP.. Bria iPhone Edition softphone app easy to use API media, some don ’ t care about chan_pjsip chan_sip...
2020 pjsip vs counterpath