Rtpproxyemplois

j'ai besoin de installer un serveur opensips avec une proxy rtp (rtpproxy) + serveur Freeswitch le tout avec debian jessie x64 comme sur le plan: Merci

Dear partners, We are looking for a RTPProxy consultant. We have RTPProxy installed in a Centos7 dedicated server and calls are sent from Sippy B2BUA system. We are getting an issue as during high load RTPProxy is dropping calls. The maximum load we are trying to get in this dedicated machine, is: 5000 ports. B2BUA and RTPProxy are running in different machines. We have tried different scenarios but it is still giving this issues after some minutes of run. We tried also running different instances, single instance and running as root user. If anyone is interested, feel free to contact me.

We need a SBC installation (Kamailio or OpenSIPS, preferably Kamailio), which will have several Asterisk servers behind, that do not use realtime (and won't use) but use a database to generate config files (both extensions and sip). Said SBC must manage the authentication, but Asterisk peers should still be independent and have their own config (things like context, call-limit...(things like context, call-limit, accountcode, etc.). Due to not using realtime for Asterisks, registration control should be managed in the SBC's own database. The SBC will have to load balance between the asterisk, and have failover between them. It should have the basic security modules as well (e.g. secfilter with GeoIP for Kamailio). Also it should be able to control NAT and RTP as well (rtpprox...

Hi, Given: SIP-Server A (IP A) SIP-Server C (IP C) OpenSIPS B (IP B) Calls can be from A->C and C>A. A->B->C C->B->A OpenSIPS 3.2 (B) in the middle should route the Calls with a RTPProxy. OpenSIPS should generate CDRs.

I have installed OpenSIPS 3.1 server with UAC, CP and rtpproxy on Debian 9 server with 2 network interface. (Asterisk server - OpenSIPS - Service Provider) Now I am able to establish calls and also registering to my service provider. I need someone's help to fix the rtp traffic and routing.

Hello, we need a kamailio expert that can configure current kamailio working on dispatcher mode, to work with rtpengine/rtpproxy. Best Regards

hi i have a ASTERISK SERVER and it have a SIP Account i have a kamailio5 installed on my local system now you need to register kamailio5 to Asterisk using SIP account when asterisk server send call to kamailio5 ==========kamailio5 have to send call to local gateway IP like if you can make it please bid

Hi There, We require help to build a proof of concept.s We want a system connect a Client to a Carrier that is scalable. However, we do not want to disclose any of the Carrier IPs to the Client and Vice Versa. I've done some research and using OpenSIPS and RTPproxy can help with this but I am having trouble setting this up on AWS. For the proof of concept, we can use 2 PABXs one to act as a Carrier, another to Act as a Client. What we would like for you to do is: 1. Configure a Linux Instance (Please provide the build and we can organise it with AWS) to: a. Connects to a Client SIP Trunk (We will use a Hosted PABX to test) b. Connects to a Carrier SIP Trunk (We will use a Hosted PABX to test) c. Passes calls between the Client and the Carrier d. Will hide the IP of both M...

OpenSIPS/Sippy Carrier Routing - Stage 1 either an OpenSIPS or Sippy box will act as an outbound traffic router for multiple carriers. Calls will be sent from already running Asterisk boxes to the OpenSIPS/Sippy/SippyGO box (we will prepare base OS, OpenSIPs/SIPPY core installs) Media needs to be proxied with RTPProxy/etc through the OpenSIPS/Sippy/SippyGO box as well. Must handle ULAW/ALAW/G729. Each outbound carrier will be assigned a prefix, which needs to be stripped out and then the traffic sent out to the respective carrier. The Asterisk Box IPs need to be able to be whitelisted at the OpenSIPS/Sippy/SippyGO box level. Non whitelisted IPs should not be able to send calls through.

opensips (или что то другое) смотрит в мир исходя из домена на какой идет подключение(смотрит в базе mysql) направляет на нужный asterisk в локальной сети. нужно настроить opensips + rtpproxy

Need opensips installation and configuration ip hiding, rtpproxy, pike modul conf., cgrates, multitenant users and groups, load balance with media(fs), class 4 features. Urgent delivery will paid extra tip.

Need opensips installation and configuration ip hiding, rtpproxy, pike modul conf., cgrates, multitenant users and groups, load balance with media(fs), class 4 features. Urgent delivery will paid extra tip.

Здравствуйте. Интересуе...предоставляет телефонию. Мы хотим интегрировать эту телефонию в CRM для приема звонков (и вызовов) из браузера. Для этого нам необходимо настроить промежуточный сервер, на котором Kamailio бы общался с браузером по Websocket и проксировал все провайдеру. Настроить Websocket у нас получилось, однако вызовы идут только в одну сторону (от браузера) и голосовой трафик не передается. Полагаю неверно настроен Rtpproxy. Жду от вас ответа. Спасибо

Need working Kamailio 4.3 configuration for a pass-through proxy with RTP relay. The pass-through SIP proxy will forward all SIP messages (including REGISTER, SUBSCRIBE, NOTIFY, OPTIONS, etc.) and also passes all RTP traffic through RTPProxy. We are using Debian 8 with the standard packaged Kamailio and RTPProxy versions. Standard sanity checking and security options (fragmented packet, ddos, etc.) should also be implemented. Allowed domains / routes will be stored in text db files for ease of maintenance / replication. The objective is allows to multiple separate back end Asterisk PBXs to use a single public IP addresses. This is not a load balancing setup. Similar project to what we are looking to accomplish, just with a far older version of Kamailio:

Looking for someone who has implemented audio/video calling in android using kamilio and RTPProxy . Please bid if you already developed and have source code .

We have custom setup of 1 Opensips as proxy and 2 Freeswitch for scripts using opensips script. Now we would like to move media / rtp process on opensips (Want to handle it using rtpproxt) rather than handing it using freeswitch. Note : Servers are behind NAT. So we might not be able to give you access of servers. You will need to provide customized opensips cfg file and steps to install rtp proxy. Please do bid only if you really did similar job previously and can do it quickly.

We have custom setup of 1 Opensips as proxy and 2 Freeswitch for scripts using opensips script. Now we would like to move media / rtp process on opensips (Want to handle it using rtpproxt) rather than handing it using freeswitch. Note : Servers are behind NAT. So we might not be able to give you access of servers. You will need to provide customized opensips cfg file and steps to install rtp proxy. Please do bid only if you really did similar job previously and can do it quickly.

I need to bypass my ISP as they are blocking port 5060, and other port for SIP. I need to know if I can bypass using proxy to connect my PBX.. Based on my research I want SIP Proxy and RTP Proxy installed on my ubuntu which will be used as proxy server to bypass there are software like reSIProcate and rtpproxy . I want some one to install and configure so it can be used Thanks.

Hi I still need someone to develope my opensips server. I attached the project details What we have done: • Installed Opensips 2.1 with default configuration • Installed ASTPP • Installed RTPProxy • We are developing Android client We want you to do these jobs: • Solve NAT traversal problem • Integrate an installed ASTPP with Opensips • Pass SIP connections from our server to the other SIP Server (PSTN Network)

...freedom! You cannot run Kamailio from behind a home network with NAT if you want to call anyone outside your home. Seriously, if you want to do this you are in for a world of advanced IP networking configuration and application code. It's possible but you really do not want to do this Kamailio -- A modular SIP router, user registration server, and NAT traversal utility...and so much more rtpproxy -- a small utility to proxy encrypted audio and video streams. Works with Kamailio to solve NAT traversal Freeswitch -- A SIP softswitch. Provides testing services like an echo test. Can also provide automated call services like voicemail. PostgreSQL -- The World's Most Advanced Open Source Database. Stores user account data and domain alias information ...

I need a VOIP Server for maximum of 1500 simultaneous calls between Android clients and a PSTN network. I installed an Opensips 2.1 server on Ubuntu Server 14.04 but have these problems: 1. Opensips Clients can call each other but without audio transition. It seems that this is a NAT traversal problem. I installed RTPProxy but could not solve the problem. 2. How can I configure this server with PSTN network? 3. How can implement a prepaid billing for my Opensips using ASTPP? I installed ASTPP but could not connect it to Opensips. I want you to configure my server and learn me how can I do this myself.

...do audio or video codec. Here is what I need from you, I need codec translation, can you use: opensip+media proxy(for example asterisk pbx). I need,m for asterisk+opensip setup opensips act as load balancer and asterisk work as media gateway. possible do asterisk only. it will be simple, but harder add another server if needed. I need from you: setup of opensips with nat traversal and rtpproxy. it is more scalable variant, but without asterisk(codec translation). You can add asterisk support(only one video codec, but audio codec translation). Get it to work for me with video using a pair of xlite i need you to setup dialling via 9xxx do via asterisk. so you can test both variants. I figure I need calling via asterisk as default, so you will change config for fr...

I need: opensip+media proxy(for example asterisk pbx). I need codec translation I need video codec and audio codec I need asterisk+opensip setup opensips act as load balancer and asterisk work as media gateway. possible do asterisk only. it will be simple, but harder add another server if needed. I need from you: setup of opensips with nat traversal and rtpproxy. it is more scalable variant, but without asterisk(codec translation). You can add asterisk support(only one video codec, but audio codec translation). i need you to setup dialling via 9xxx do via asterisk. so you can test both variants. I figure I need calling via asterisk as default, so you will change config for free. there almost no additional complexity. asterisk is pbx. opensips is sip proxy. asteri...

Need to implement RTP header compression in Linphone/oRTP and in Opensips/RTPproxy.

I am in the process to configure Kamailio with Asterisk for personal use. Below are the setup deta...setup details: - Asterisk and Kamailio on one VPS with 1 public IP - Asterisk listening at Port XXXX - Kamailio is listening at port YYYY - No P2P calling Kamailio is acting as registrar and Location server at the moment. INVITES are being forwarded to asterisk by Kamailio. I AM FACING THE NAT ISSUES BECAUSE THE EXTENSIONS ARE BEHIND NAT. ONE WAY AUDIO OR NO AUDIO AT ALL WHEN CALLING. Tried rtpproxy but it seems I am not the expert of configuring routes in Kamailio so I am looking for someone who can help me to fix and understand the issues. Budget: Setup is for my personal use so I don't have any budget in my mind but if you can tell me how much it would cost then we can ...

I am in the process to configure Kamailio with Asterisk for personal use. Below are the setup deta...setup details: - Asterisk and Kamailio on one VPS with 1 public IP - Asterisk listening at Port XXXX - Kamailio is listening at port YYYY - No P2P calling Kamailio is acting as registrar and Location server at the moment. INVITES are being forwarded to asterisk by Kamailio. I AM FACING THE NAT ISSUES BECAUSE THE EXTENSIONS ARE BEHIND NAT. ONE WAY AUDIO OR NO AUDIO AT ALL WHEN CALLING. Tried rtpproxy but it seems I am not the expert of configuring routes in Kamailio so I am looking for someone who can help me to fix and understand the issues. Budget: Setup is for my personal use so I don't have any budget in my mind but if you can tell me how much it would cost then we can ...

Bid ONLY if you are extremely skilled with OpenSIPs, Asterisk, RTPProxy, MySQL. Also bid only if you are available to work all day Sunday (your timezone). I had a speed testing application created for me which was housed on my server where users would download a softphone and have it running in their system tray. The user softphone registers to opensips. Every 15 minutes, Asterisk will call this softphone and measure jitter, latency, upload/download speed. The results will be inserted into MySQL. This system was and still is working perfectly. The change I would like you to make is instead of having the server not being behind a router (as the original developer had suggested), I now need the server to be behind the router. You would need to read through a 5 page documen...

Kamailio (+rtpproxy) используется как sip-proxy для регистрации sip-клентов, находящихся в публичной сети (Интернет), и обеспечивает сквозную регистрацию на asterisk (через общую базу). Asterisk находится на другом сервере. Kamailio специально изпользуется как бордер, чтобы не подвергать риску сервер с Asterisk, на нем регистрируются клиенты только с приватных сетей. Основная проблема - клиенты за двойным натом, в результате, как правило, односторонняя слышимость, иногда нет слышимости и иногда нет регистрации. При переключении данного абонента напрямую на Asterisk - проблем нет. Требуется провести аудит данного решения, проверить конфиги kamailio, rtpproxy и asterisk на предмет взаимодействия.

Abstract. Request for an complete Kamailio-sipserver install documented...changes in cfg and startfiles documented. STUN (If needed): · All apt-get installed documented. · All makes installed and documented. · All changes in cfg and startfiles documented. NAT / Nathelper (If needed): · All apt-get installed documented. · All makes installed and documented. · All changes in cfg and startfiles documented. RTPproxy (If needed): · All apt-get installed documented. · All makes installed and documented. · All changes in cfg and startfiles documented. Verification: The connection between two sip users connected to our server s...

We are looking for someone to support and optimize our Opensips+RTPproxy load balancer instance. Tasks include testing and optimizating the server, providing feedback on how it can be optimized. The load balancer sends calls to multiple Asterisk nodes which do the billing and routing. The candidate must be an expert on and should have proven knowledge of Opensips/Kamailio and RTPproxy.

I'm looking to integrate Asterisk, Kamailio, rtpproxy and a customer self-service interface. Kamailio will only be used for proxy and routing. Asterisk will be used for all media services, including voicemail, conferencing and IVR. I will provide you with the following: 1 Server for Kamailio + rtpproxy 2 Servers for Asterisk 1 server for MySQL and Apache/nginx I need the following Kamailio installed on one server, communicating with MySQL, providing Registration, Extension to Extension calling and Multi-Domain services Asterisk installed on one additional server providing both Conferencing and IVR Asterisk installed on one additional server providing Voicemail Services communicating with MySQL via ODBC for configuration and Voicemail Storage MySQL and a Web...

Required in Bangalore freelance VoIP Engineer over 4 years experience in C++, SIP, PJSIP, Symbian, Carbide, Android, Kamailio (open SER), RTPProxy, Good understanding of Media codecs, RTP etc will be an advantage. Flexible timing available and work will be on project basis.

...the middle of all SIP dialogs and properly forward all types of requests up and downstream 6. Inbound calls from SIP provider should not be challenged for password. IP based authorization only 7. Sippy will try to detect idle connections (ie lost BYE packet) by making re-invite and it also needs to be handled 8. Only Sippy is responsible for rewriting SDP and deciding to use or not use RtpProxy. No RtpProxy handling on OpenSIPS side is needed Desired call flows: Outbound call (remote SIP client) -> OpenSIPS -> Sippy -> SIP provider Inbound call (SIP provider) -> OpenSIPS -> Sippy -> remote SIP client (user registered to OpenSIPS) Detailed issues with current config: 1. 404 Not Here error for BYE and CANCEL requests. Has_totag returns true but ...

Hi, I am looking for the exact same solution. Would you please do it for me?? http://www.freelancer.com/projects/Script-Install-Linux/OpenSIPS-rtpproxy-transperent-router.html Thanks in advance. Regards

We have Softswitches behind NAT. You need to setup Kamailio with RTPPROXY/Mediaproxy just to forward the SIP traffic to my softswitches. No authentication or whatsoever needed. Just routing SIP Trafficproperly through the NAT. You might also need to reconfig , iptables etc to synchronize the whole thing. You should discuss with us what you did with Kamailio for our future addition/change to new sip servers. At the end of project sip connections with real sip trunk will be tested. Someone who knows about Kamailio, can finish it in 15 mins. Before you begin you will get our network schematic which is pretty simple and you should provide us with tipps about where to add the Kamailio server. I Appreciate your response!

Similar to the project that you did before, I need an opensips/rtpproxy server configured to proxy communication between my DMZ and internal network.

I would need you to install and configure openSIPS on our Debian 6.0 server running on Amazon EC2. It should be configured so that: traffic from devices behind a symmectrical NAT would have their RTP routed through either a rtpproxy or mediaproxy on that server. You would also need to install the rtp proxy and cofigure it accordingly. Traffic from all other devices should be handled by the nathelper/nat_traversal modules so that we can avoid routing the traffic through our system. Additionally I would like the openSIPS to be able to use an external mySQL DB for providing user credentials. I would need proper documentation of all installation and configuration steps taken. Time is (as usual... ;) ) of the essence so workers being able to start asap will be preferre...

...asterisk 1.8 Running on Amazon EC2 But i have issues with nat clients so i'm thinking to use Kamailio to solve nat issues and also get a more scalable system. I can give you Teamviewer access. I need the following: We need a serious professional to install and configure Kamailio integration with our Asterisk + A2Billing platform. - Kamailio should act as Registrar with Nathelper + RTPProxy/MediaProxy and it should allow local users to call each other without passing through Asterisk. It should also route local, national and international calls like to Asterisk + A2Billing so i can still bill them. - Kamailio should authenticate users using the Asterisk/A2Billing database - Asterisk + A2Billing should be configured so that new user accounts would be create...

...asterisk working in realtime. But i have issues with nat clients so i'm thinking to use Openser to solve nat issues and also get a more scalable system. I will give you Teamviewer access. So i need the following: We need a serious professional to install and configure an OpenSER (Kamailio) integration with our Asterisk + A2Billing platform. - OpenSER should act as Registrar with Nathelper + RTPProxy/MediaProxy and it should allow local users to call each other without passing through Asterisk. It should also route local, national and international calls like (011/00 prefix) to Asterisk + A2Billing. OpenSER should use a View of the A2Billing database to authenticate users. - Asterisk + A2Billing should be configured so that new user accounts would be created in A2...

I need OpenSIPS 1.6.4 configuration that will just transparently route to our Asterisk 1.6.2.x Server every SIP connection it gets (also you need to use rtpproxy). No DB backend. No OpenSIPS authentication of peer. Just routing of SIP and RTP traffic to Asterisk. Your job is to make a working file to route SIP traffic between a SIP UA and Asterisk with the help of rtpproxy. SIP UA should connect to the IP of OpenSIPS, and OpenSIPS should route everything transparently with the help of rtpproxy to Asterisk. Why I need this? Because due to some routing problems I'm unable to connect some client phones in some cities to our main Asterisk server. I've setup a server that is visible to those clients but I want it to be only a SIP router. I will provide ...

I need help with OpenSIPS setup. I have already installed OpenSIPS with rtpproxy, but there is some problems with internal (from one to other user registered on this sips but transfered by asterisk to transcode) - there is one way audio problem.

...--Internal calling user to user + International Calling . --Alias with DID Maping --PSTN ( over SIP)calling/ billing --DID forwarding --Call forwarding --Call forwarding on busy --Call forwarding on no answer --Voicemail- Simple Text Reader +(translation). --PIN request. --LCR(Least Cost Routing) / Carrier route --Enum Support --Video calling should support. --NAT traversal support (RTPproxy, Mediaproxy or same like SBC) --Text Messaging --Webphone --Callthrough --Click to Call (Callback) --IVR (Tell the Balance every time user call). --Web Call Back 1 Customer enters source and destination numbers in Click to Call back web page. 2 System calls source number 3 When user picks up the phone, he is connected to destination number 4 Ability to activate/deac...

I am re-posting this project due to failure from provider. We need a serious professional to install and configure an OpenSER (Kamailio) + Asterisk + A2Billing platform. - OpenSER should act as Registrar with Nathelper + RTPProxy/MediaProxy and it should allow local users to call each other without passing through Asterisk. It should also route international calls (011/00 prefix) to Asterisk + A2Billing. OpenSER should use a View of the A2Billing database to authenticate users. - Asterisk + A2Billing should be configured so that new user accounts would be created/added in A2Billing. Also, each user should have their own voicemail account and OpenSER should route unanswered/failed calls to the user's voicemail. - Wholesale inbound calls should be authenticated by IP...

We need a serious professional to install and configure an OpenSER (Kamailio) + Asterisk + A2Billing platform. - OpenSER should act as Registrar with Nathelper + RTPProxy/MediaProxy and it should allow local users to call each other without passing through Asterisk. It should also route international calls (011/00 prefix) to Asterisk + A2Billing. OpenSER should use a View of the A2Billing database to authenticate users. - Asterisk + A2Billing should be configured so that new user accounts would be created in A2Billing. Also, each user should have their own voicemail account and OpenSER should route unanswered/failed calls to the user's voicemail. - Wholesale inbound calls should be authenticated by IP address.

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...over SIP)calling/ billing * DID forwarding * Call forwarding * Call forwarding on busy * Call forwarding on no answer * Voicemail(Using feature server Asterisk/SEM as voicemail server) - I need just a simple configuration, with new text, and voice file from me (translation). PIN request. * LCR(Least Cost Routing) / Carrier route * Enum Support * Video calling * NAT traversal support (RTPproxy, Mediaproxy or same like SBC) This is the pretentious and sophisticated part on the configuration. Please bid if you are a NAT traversal expert. Please tell me what kind of solution you want to use. * Text Messaging - should be able to use openWengo - Jabber XMPP Voice, IM support - Webphone * Callthrough * Click to Call (Callback) 1 Customer enters source and destination...

I need OpenSER 1.2 configuration that will just transparently route to our Asterisk Server every SIP connection it gets (also you need to use rtpproxy). No DB backend. No OpenSER authentication of peer. Just routing of SIP and RTP traffic to Asterisk. So your job is to make me a working file to route SIP traffic between SIP client and Asterisk with a help of rtpproxy. SIP client should connect to IP of OpenSER and OpenSER should route everything transparently with a help of rtpproxy. Why I need that? Because due to some routing problems I'm unable to connect some client phones in some cities to our main Asterisk server. I've setup a server that is visible to those clients but I want it to be only a SIP router. I will provide client with alternati...

Urgently need OpenSER 1.2 configuration file that will just transparently route to our Asterisk Server every SIP connection it gets(it will also need to use rtpproxy). Every manual in web I found shows me how to autenthicate SIPs in OpenSER using MySQL. But I don't need that. So your job is to make me a working file to forward route SIP traffic between SIP client and Asterisk. SIP client should connect to IP of OpenSER and OpenSER should route everything transparently with a help of rtpproxy. Why I need that? Because due to some routing problems I'm unable to connect some client phones in some cities to our main Asterisk server. I've setup one more server with different routing but I want it to be only a router. I will provide client with alternativ...

...radius modules. Subscribers are stored in a MySQL database. The subscriber id should be an email address in the form user [at] domain. Should assume that subscribers exists in the following table: CREATE TABLE users ( name varchar(50), domain varchar(100), passwd varchar(40) sipenabled bool, // if false then user is not granted permission to the service ); NAT traversal support. STUN, RTPproxy, nathelper should be installed and configured for NAT/firewall. You should test your configuration using openwengo 2.1. Openwengo calls, chats and presence (using SIP/SIMPLE) should work. You should also configure openSER to route off-net phone calls through a sipphone account. For example, if the user dials a number starting with "00" the call should be performed thro...