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    1,059 linux g729 tugasan ditemui, harga dalam USD

    i have voip device and i have small router it have 32mb ram os Openwrt i want to run 32calls so you need to use SIP2...best libre [login to view URL] or yate or besip [login to view URL] up to you as you like i just need pass a call on g729 codec under 15KBPS now as you like to do i have no prob

    $216 (Avg Bid)
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    3 bida

    I have a Fusion pbx but need G729 codec loaded

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    14 bida

    I nee...user/pass or no authentication. - Make calls with or without authentication - Forward DTMF digits using RFC2833 or inband - Use codec G711 and GSM - Be able to use codec G729 - Receive through GSM one 1 simultaneous call and rout it to a predefined SIP client - Be able to store a CSV with all calls made Skills required: Android, VoIP

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    I need Sip Dialer With Include OpenVPN. Sip Dialer connect by OpenVPN. OpenV...Include OpenVPN. Sip Dialer connect by OpenVPN. OpenVPN Connect only Sip dialer..... you need use PJSIP OpenSource SDK for SIP Dialer. I need Very Simple Apps... i need support g729 voice code. I can you OpenVPN server side. if you have good experience please contact me.

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    5 bida

    I need android VoIP Softphone With Include VPN .. You need use PJSIP OpenSource SDK for Softphone.... You can use any VPN technology for softphone. I need Support g729 voice code. I need Good Voice Quality. If you have Old Experience for Softphone with vpn then contact me.

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    I need to compile the linphone softphone and let it work with its codec g729, so I can work it in an android studio with windows, after compiling the use linphone account button I require that the username and password be sent to the [login to view URL] domain [login to view URL] Thank you very much

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    I ne...customize it, I require after compiling the open source code that the app is registered using only username and password, the app must be configured to support the codecs at least g729, g711, g722, gsm, the person who wants to work I sent the attached source code so you can compile it and send me the demo to start working Thank you very much

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    ...just need to initiate a call on gsm network using core android functions and audio will be automatically forwarded. Maybe this is a solution. Codecs: Minimum g711, desired g729. Full description below. Please review and message me with considerations. If any requirements are unable to be met, please let me know before we make a deal. Summary

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    I need to sync rtp with caller and called with RTCP SR. Project must be done in c# . SSRC and Payload changes. Codecs are G711,G729,AMR WB,AMR NB,EVS.

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    ...IP, user/pass or no authentication. - Make calls with or without authentication - Forward DTMF digits using RFC2833 or inband - Use codec G711 and GSM - Be able to use codec G729 - Receive through GSM one 1 simultaneous call and rout it to a predefined SIP client Skills required: Android, Asterisk PBX, VoIP An standard SIP client running on a seperate

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    Update ODOO Tamat left

    Transfert ODOO from current in-house server to a new server hosted in a UK based provider to be chosen together ( datacenter ...invoice, prevent from generating proforma invoice in Europe without VAT number) Install Elastix IPBX on same server and test a call (we will provide telecom trunk), install g729 codec Move 1 wordpress website to that server

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    requiero un softphone para a2billing como el de movisnow que permito codec g729

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    ...should work for both and each by separate registration. 4- send calls from sip/iax2 to gsm and receive calls from gsm and send it to sip/iax2 5- Support codecs G711a , G711u, G729, gsm, slin and OPUS. 6- ability to change the mobile IMEI and mobile MAC address on request by API, 7- handle USSD codes and get all type of responses (popups, notifications

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    ...------------------------------- 1. user login by sip server user 2. show balance display 3. show call log 4. support incoming & Outgoing Call 5. Working Smooth 6. support g729,g729u & others voice code 7. Include any VPN Technology for SIP Block Country 8. Smart Graphical Interface 9. Include AdMob Ads **** When Complete Project then need provide

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    ...have all call features and locally store call details on device like dial/missed/received call. Softphone must be able to support audio and vedio calls with suitable codec. G729 is must for this. Users will be able to add multiple SIP accounts. if you already developed this kind of soft-phone, share the demo link. if you are from Meerut/Delhi or

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    ...traditional GMS getaway) This mobile application will work on only wifi Internet connectivity, coz GSM internet data normally disable during any GSM call. All call will pass with G729 codec The registration server may be Asterisk or VOIP switch or any other server or customize server. This server will receive call from another VOIP switch server with SIP

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    ...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

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    ...need it for our clients to enable them to register with our Asterisk Softswitch and order to make and receive VoIP calls while on GPRS, Edge, 3G WIFI etc. The codec should be g729 or similar. We need the clients to download the pre-configured software to their phones, enter their account name and password, and they should be able to make/receive calls

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    ... so it will work like DIALER(DIALS NUMBER)--> MY SOFTSWITCH --> ANDROID PHONE --> TO NUMBER DIALED BY DIALER.... This will be main future.. and it will work on codecs g711 g729 g723 ____________________________________________________________________________________________________________________________________________ Other then this i want

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    ... so it will work like DIALER(DIALS NUMBER)--> MY SOFTSWITCH --> ANDROID PHONE --> TO NUMBER DIALED BY DIALER.... This will be main future.. and it will work on codecs g711 g729 g723 ____________________________________________________________________________________________________________________________________________ Other then this i want

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    Mobile Dialer Tamat left

    Mobile VoIP Dialer for IPPBX Features required: Call to extension or ca...Features required: Call to extension or call to a number or contacts Integration with Mobile Phonebook Call Transfer Call Hold Call conference ./ multiparty conference call G729, GSM, iLBC, G711, G722, AMR Codec support Tunneling will be considered as additional value

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    Mobile Dialer Tamat left

    Mobile VoIP Dialer for IPPBX Features required: Call to extension or ca...Features required: Call to extension or call to a number or contacts Integration with Mobile Phonebook Call Transfer Call Hold Call conference ./ multiparty conference call G729, GSM, iLBC, G711, G722, AMR Codec support Tunneling will be considered as additional value

    $517 (Avg Bid)
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    ... so it will work like DIALER(DIALS NUMBER)--> MY SOFTSWITCH --> ANDROID PHONE --> TO NUMBER DIALED BY DIALER.... This will be main future.. and it will work on codecs g711 g729 g723 I KNOW ITS NOT POSSIBLE IN ANDROID BUT MANY PEOPLES CONFIRMED THAT ITS POSSIBLE IN ANDROID WITH CHIPSET MTK6577. ____________________________________________________

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    ...traditional GMS getaway) This mobile application will work on only wifi Internet connectivity, coz GSM internet data normally disable during any GSM call. All call will pass with G729 codec The registration server may be Asterisk or VOIP switch or any other server or customize server. This server will receive call from another VOIP switch server with SIP

    $10 (Avg Bid)
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    1 bida

    ...so it will work like DIALER(DIALS NUMBER)--> MY SOFTSWITCH --> ANDROID PHONE --> TO NUMBER DIALED BY DIALER.... This will be main future.. and it will work on codecs g711 g729 g723 I KNOW ITS NOT POSSIBLE IN ANDROID BUT MANY PEOPLES CONFIRMED THAT ITS POSSIBLE IN ANDROID WITH CHIPSET MTK6577. ________________________________________________

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    ... so it will work like DIALER(DIALS NUMBER)--> MY SOFTSWITCH --> ANDROID PHONE --> TO NUMBER DIALED BY DIALER.... This will be main future.. and it will work on codecs g711 g729 g723 I KNOW ITS NOT POSSIBLE IN ANDROID BUT MANY PEOPLES CONFIRMED THAT ITS POSSIBLE IN ANDROID WITH CHIPSET MTK6577. ____________________________________________________

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    I need to use an android cellphone as an...user/pass or no authentication. - Make calls with or without authentication - Forward DTMF digits using RFC2833 or inband - Use codec G711 and GSM - Be able to use codec G729 - Receive through GSM one 1 simultaneous call and rout it to a predefined SIP client - Be able to store a CSV with all calls made

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    24 bida

    I need to use an android cellphone as an...user/pass or no authentication. - Make calls with or without authentication - Forward DTMF digits using RFC2833 or inband - Use codec G711 and GSM - Be able to use codec G729 - Receive through GSM one 1 simultaneous call and rout it to a predefined SIP client - Be able to store a CSV with all calls made

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    Project Description I need an applicati...user/pass or no authentication. - Make calls with or without authentication - Forward DTMF digits using RFC2833 or inband - Use codec G711 and GSM - Be able to use codec G729 - Receive through GSM one 1 simultaneous call and rout it to a predefined SIP client - Be able to store a CSV with all calls made

    $526 (Avg Bid)
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    8 bida

    I need to use an android cellphone as an...user/pass or no authentication. - Make calls with or without authentication - Forward DTMF digits using RFC2833 or inband - Use codec G711 and GSM - Be able to use codec G729 - Receive through GSM one 1 simultaneous call and rout it to a predefined SIP client - Be able to store a CSV with all calls made

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    ...60-80% in reality then theory) from Server X to Server Y. A usual SIP in G729 call takes 27-32kbps per port Explanation of scenario: 1. Server X (asterisk server, with static IP) receiving VoIP calls from different Carrier/Originator, with h323/sip protocol, using G711,G729 and/or G723r6.3 codec and sending calls to Server Y. 2. Server Y {Windows

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    ...Asterisk Server Server B = Asterisk Client Explanation of Scenario: 1. Server A ( asterisk server, with static IP) receiving VoIP calls , with sip protocol, using G711,G729 and/or G723.1 codec and sending calls to Server B 2. Server B ( Asterisk server with PRIVATE NETWORK IP), receiving calls from server A and sending to gateways (quintum gateway

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    I want to add codec G729 to increase quality call on my system and make a modification on my server, check my setting to ensure i got them right. Currently cannot make international calls, suspect codec as my providers use G729, G711, G711u etc and not ulaw or alwaw which are default for a2billing. A few small bits here and there.

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    ...network. (this part just like a traditional GMS getaway) This mobile application will work on wifi Internet connectivity and mobile Data internet , All call will pass with G729 codec The registration server may be Asterisk or VOIP switch or any other server or customize server. This server will receive call from another VOIP switch server with SIP

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    24 bida
    SIP Client Tamat left

    SIP Client for IOS с поддержкой кодека G729

    $249 (Avg Bid)
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    1 bida

    ...network. (this part just like a traditional GMS getaway) This mobile application will work on wifi Internet connectivity and mobile Data internet , All call will pass with G729 codec The registration server may be Asterisk or VOIP switch or any other server or customize server. This server will receive call from another VOIP switch server with SIP

    $603 (Avg Bid)
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    10 bida

    ...traditional GMS getaway) This mobile application will work on only wifi Internet connectivity, coz GSM internet data normally disable during any GSM call. All call will pass with G729 codec The registration server may be Asterisk or VOIP switch or any other server or customize server. This server will receive call from another VOIP switch server with SIP

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    2 bida

    A Sip Dialer App is required with g729 codec for iPhone and Android. App should have functionality to change settings for codec, jitter, communication ip/port and add/remove account setting for customer.

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    im looking for a soft phone application for desktop, android -auto provision -contact -good quality sound -low bandwitdth G729 H264 -TLS / SSL -all business features ( trans, conference,hold, mute........ ) VOIP- ANTIBLOCK IP AND TRAFFIC COMPRESSION solutions for censored countries such as egypt, Iran, China call center soft ware with

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    gsm gateway Tamat left

    ... so it will work like DIALER(DIALS NUMBER)--> MY SOFTSWITCH --> ANDROID PHONE --> TO NUMBER DIALED BY DIALER.... This will be main future.. and it will work on codecs g711 g729 g723 I KNOW ITS NOT POSSIBLE IN ANDROID BUT MANY PEOPLES CONFIRMED THAT ITS POSSIBLE IN ANDROID WITH CHIPSET MTK6577. ____________________________________________________

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    gsm gateway Tamat left

    ... so it will work like DIALER(DIALS NUMBER)--> MY SOFTSWITCH --> ANDROID PHONE --> TO NUMBER DIALED BY DIALER.... This will be main future.. and it will work on codecs g711 g729 g723 I KNOW ITS NOT POSSIBLE IN ANDROID BUT MANY PEOPLES CONFIRMED THAT ITS POSSIBLE IN ANDROID WITH CHIPSET MTK6577. ____________________________________________________

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    I nee...user/pass or no authentication. - Make calls with or without authentication - Forward DTMF digits using RFC2833 or inband - Use codec G711 and GSM - Be able to use codec G729 - Receive through GSM one 1 simultaneous call and rout it to a predefined SIP client - Be able to store a CSV with all calls made Skills required: Android, VoIP

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    14 bida

    ... so it will work like DIALER(DIALS NUMBER)--> MY SOFTSWITCH --> ANDROID PHONE --> TO NUMBER DIALED BY DIALER.... This will be main future.. and it will work on codecs g711 g729 g723 I KNOW ITS NOT POSSIBLE IN ANDROID BUT MANY PEOPLES CONFIRMED THAT ITS POSSIBLE IN ANDROID WITH CHIPSET MTK6577. ____________________________________________________

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    ...attach more huawei usb modems to the server if needed. USB hubs will be used to attach the usb modems. - The server shall be able to terminate voip calls comming in codecs g729 and g723. - Send ussd and ivr to each sim card to topup and credit check. - Display how much credit is available in each sim card. - be able to listen to the channels

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    Hello, need someone expert to install C5 Switch and doing this parts: Enable Voice and video calling Enable G729 Codec And add SIP Trunk

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    Trophy icon Convert pcap file to MP4 Tamat left

    An application that can convert a pcap file to MP4. pcap file has RTP data in g711,g729, H264 codes. G711 and G729 for voice and H264 for Video. Application should have no UI, just have a command line option to pass path for pcap file. the output MP4 should be named with same name as pcap. eg if we pass c:folder1[login to view URL] it should create C:folder1123

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    Dijamin
    $250
    4 penyertaan

    Expert in C, C++, PJSIP Stack Must be able to analyze the Wireshark captures to identify any SIP signaling/media issues. Should have worked on Server side S...analyze the Wireshark captures to identify any SIP signaling/media issues. Should have worked on Server side Should be aware of RFC 3261,3264, NAT traversal, Media Codecs AMR, G729, Opus.

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    3GPP TS 33.108 V.10.3.0. ASN.1 This project is for a software to open ASN 1 [login to view URL] 1 [login to view URL] and decrypt audio part with this codec Voce: AMR WB, AMR NB, G722, G711, G729 • Fax: T38, G711 • Video: H263, H263+, H264avc Decode

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    Hello guys, This project is belong to who make are expert. i need to install FreeSwitch on my server and install ASTTP on that. then doing this configs: 1- Activate G729 Codec on that 2- Change transfer protocol to UDP 3- activate just SIP protocol on server. (disable another protocols) 4- Enable Video Call feature on freeswich if you

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    REQUESTS: 1- compile G729 codec for asterisk 1.8 on mips device 2- compile new version of FFMPEG because old version not work with online link example: root@arc-pbx:~# ffmpeg -i [login to view URL] ...... libavutil version: 49.6.0 libavcodec version: 51.55.0 libavformat version: 52.13.0 libavdevice version: 52

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