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    152 ilbc linux jobs fundet, i prisklassen EUR

    ...Integration with the native android contact list - Speakerphone mute and hold - UDP and TCP transports (use TCP for better battery life!) - Supports g711 (ulaw, alaw), speex, iLBC and gsm codecs - Supports sending of DTMF - DNS SRV - Built-in echo cancellation - STUN support - Change ringtone per account - call waiting - Video support (gold users only)

    €2261 (Avg Bid)
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    18 bud
    Mobile Dialer Udløbet left

    Mobile VoIP Dialer for IPPBX Features required: Call to extension or call to a n...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

    €393 (Avg Bid)
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    Mobile Dialer Udløbet left

    Mobile VoIP Dialer for IPPBX Features required: Call to extension or call to a n...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

    €455 (Avg Bid)
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    15 bud

    Call functions like mute, confe...integration from LDAP, Outlook, or CSV Low resource consumption Supports SIP, XMPP, and IAX accounts TLS, SRTP, and ZRTP encryption Voice Codecs: G.729 (paid), a-law, U-law, GSM, iLBC 20 and 30, Speex Narrow Voice Codecs: H264 (paid), VP8 RAW SIP technology, not WebRTC. Similar Projects: Zoiper SIP Webphone

    €471 (Avg Bid)
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    Call functions like mute, conference, hold, transfer, and call rec...integration from LDAP, Outlook, or CSV Low resource consumption Supports SIP, XMPP, and IAX accounts TLS, SRTP, and ZRTP encryption Voice Codecs: G.729 (paid), a-law, U-law, GSM, iLBC 20 and 30, Speex Narrow Voice Codecs: H264 (paid), VP8 Similar Projects: Zoiper SIP Webphone

    €453 (Avg Bid)
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    I have a IP PBX in cloud and need somebody to developer a web phone app for Chrome and Firefox to access it. The app will access the IPPBX with codec G.728 and iLBC. The users only will need provide the login account and the password. The others parameters like host, proxy and network setings and audio setings will need be pre-loaded.

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    ...calls through a series of outbound carriers, as instructed by billing server Support codecs: • G711 u-law • G711 a-law • G729 • G729b • G723 • G723a • G726 • GSM • iLBC Web interface should show current activity and CDR, which should be searchable by date/time, originating IP, destination number, destination country, or outbound c...

    €1290 (Avg Bid)
    Fremhævet Haster
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    ...& RTCP (RTP Control Protocol) 8. Transport Mode TCP, User Datagram Protocol (UDP), TCP to UDP Interworking 9. Codecs G.723, G.729, AMR (Adaptive Multi-Rate Compression), iLBC, G.711, GSM. 10. Network Address Translation (NAT) Traversal Through Encryption & STUN 11. Voice Quality Silence Suppression, Packets Loss Management 12. DTMF - (Dual Tone

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    ...& RTCP (RTP Control Protocol) 8. Transport Mode TCP, User Datagram Protocol (UDP), TCP to UDP Interworking 9. Codecs G.723, G.729, AMR (Adaptive Multi-Rate Compression), iLBC, G.711, GSM. 10. Network Address Translation (NAT) Traversal Through Encryption & STUN 11. Voice Quality Silence Suppression, Packets Loss Management 12. DTMF - (Dual Tone

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    ...Protocol) & RTCP (RTP Control Protocol) 8. Transport Mode TCP, User Datagram Protocol (UDP), TCP to UDP Interworking 9. Codecs G.723, G.729, AMR (Adaptive Multi-Rate Compression), iLBC, G.711, GSM. 10. Network Address Translation (NAT) Traversal Through Encryption & STUN 11. Voice Quality Silence Suppression, Packets Loss Management 12. DTMF - (Dual Tone

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    ... Call encryption with SRTP and ZRTP for XMPP and SIP Call encryption with SRTP and SDES for XMPP and SIP CODECS Audio: Opus, SILK, Speex, G.722, PCMU/PCMA (G.711), iLBC, GSM, G.729 Annex C (requires compilation and licenses) Video: H.264, H.263-1998 / H.263+, (VP8 coming soon…) MISCELLANEOUS On-line provisioning Provisioning server discovery

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    ...& RTCP (RTP Control Protocol) 8. Transport Mode TCP, User Datagram Protocol (UDP), TCP to UDP Interworking 9. Codecs G.723, G.729, AMR (Adaptive Multi-Rate Compression), iLBC, G.711, GSM. 10. Network Address Translation (NAT) Traversal Through Encryption & STUN 11. Voice Quality Silence Suppression, Packets Loss Management 12. DTMF - (Dual Tone

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    ...& RTCP (RTP Control Protocol) 8. Transport Mode TCP, User Datagram Protocol (UDP), TCP to UDP Interworking 9. Codecs G.723, G.729, AMR (Adaptive Multi-Rate Compression), iLBC, G.711, GSM. 10. Network Address Translation (NAT) Traversal Through Encryption & STUN 11. Voice Quality Silence Suppression, Packets Loss Management 12. DTMF - (Dual Tone

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    Hey I'm running CSipSimple with codec ilbc but no audio (outgoing/incoming) can be heard in android 5.0. It works fine in android 4.4.2 with the same app. So.I need a VM file that include all development environment. I will pay USD$150 for it.

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    ...Protocol) & RTCP (RTP Control Protocol) 8. Transport Mode TCP, User Datagram Protocol (UDP), TCP to UDP Interworking 9. Codecs G.723, G.729, AMR (Adaptive Multi-Rate Compression), iLBC, G.711, GSM. 10. Network Address Translation (NAT) Traversal Through Encryption & STUN 11. Voice Quality Silence Suppression, Packets Loss Management 12. DTMF - (Dual Tone

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    ...Protocol) & RTCP (RTP Control Protocol) 8. Transport Mode TCP, User Datagram Protocol (UDP), TCP to UDP Interworking 9. Codecs G.723, G.729, AMR (Adaptive Multi-Rate Compression), iLBC, G.711, GSM. 10. Network Address Translation (NAT) Traversal Through Encryption & STUN 11. Voice Quality Silence Suppression, Packets Loss Management 12. DTMF - (Dual Tone

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    ...G711u, GSM, G722 and G729 (iLBC is desired) * DTMF (In-band, RFC 2833 and SIP INFO) * Features Mute, hold, re-dial, Transfer * 3-way conference * Multiple profiles, multiple SIP server registrations (up to 30) * XML querying of user balance * Fully compatible with Asterisk or Opensips * Windows only (No MAC or Linux needed) * Can use any free

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    ...& RTCP (RTP Control Protocol) 8. Transport Mode TCP, User Datagram Protocol (UDP), TCP to UDP Interworking 9. Codecs G.723, G.729, AMR (Adaptive Multi-Rate Compression), iLBC, G.711, GSM. 10. Network Address Translation (NAT) Traversal Through Encryption & STUN 11. Voice Quality Silence Suppression, Packets Loss Management 12. DTMF - (Dual Tone

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    ...Protocol) & RTCP (RTP Control Protocol) 8. Transport Mode TCP, User Datagram Protocol (UDP), TCP to UDP Interworking 9. Codecs G.723, G.729, AMR (Adaptive Multi-Rate Compression), iLBC, G.711, GSM. 10. Network Address Translation (NAT) Traversal Through Encryption & STUN 11. Voice Quality Silence Suppression, Packets Loss Management 12. DTMF - (Dual Tone

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    ...& RTCP (RTP Control Protocol) 8. Transport Mode TCP, User Datagram Protocol (UDP), TCP to UDP Interworking 9. Codecs G.723, G.729, AMR (Adaptive Multi-Rate Compression), iLBC, G.711, GSM. 10. Network Address Translation (NAT) Traversal Through Encryption & STUN 11. Voice Quality Silence Suppression, Packets Loss Management 12. DTMF - (Dual Tone

    €1932 (Avg Bid)
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    ...2833/4733 / RTP Inband); - Notification Center Support for missed calls, voicemail and registration status; - Blocked Numbers List. • Codecs: - Included: G.711 (uLaw & ALaw), iLBC; - Purchasable: G.722 (Wideband), iSAC (HD), G.729 Annex-B; - Support for UDP and TCP. • Other: - Call Waiting / 2nd Call / Call Toggle / Call Transfer / Conference; -

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    We are looking to integrate PortSIP SDK into our mobile app. Phase I 1. 1 to 1 Audio Call Support G.729, Opus, AMR, AMR-WB, G.722.1, G.711a/u, GSM, iLBC, SPEEX, SPEEWX-WB codecs 2. 1 to 1 Video Call Support H.263, H.263+, H.264 and VP8 video call 3. Call Transfer 4. Call Hold 5. Call Waiting 6. Call History 7. Presence Phase II 8

    €2094 (Avg Bid)
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    ...Mobile: Symbian (ALL), iPhone & iPod, Windows, Android OS, (communication via either GPRS, EDGE, 3G or Wi-Fi connection), Voice & Video 6. Dialer supports G729, G.711, G723.1, iLBC codec for sending audio & video data. 7. Dialer should be able to run behind NAT or on private IP 8. Full source code must be provided with proper comments on the code.

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    6 bud

    ...Mobile: Symbian (ALL), iPhone & iPod, Windows, Android OS, (communication via either GPRS, EDGE, 3G or Wi-Fi connection), Voice & Video 6. Dialer supports G729, G.711, G723.1, iLBC codec for sending audio & video data. 7. Dialer should be able to run behind NAT or on private IP 8. Full source code must be provided with proper comments on the code.

    €1621 (Avg Bid)
    Haster
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    ...Protocol) & RTCP (RTP Control Protocol) 8. Transport Mode TCP, User Datagram Protocol (UDP), TCP to UDP Interworking 9. Codecs G.723, G.729, AMR (Adaptive Multi-Rate Compression), iLBC, G.711, GSM. 10. Network Address Translation (NAT) Traversal Through Encryption & STUN 11. Voice Quality Silence Suppression, Packets Loss Management 12. DTMF - (Dual Tone

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    12 bud

    ...Protocol) & RTCP (RTP Control Protocol) 8. Transport Mode TCP, User Datagram Protocol (UDP), TCP to UDP Interworking 9. Codecs G.723, G.729, AMR (Adaptive Multi-Rate Compression), iLBC, G.711, GSM. 10. Network Address Translation (NAT) Traversal Through Encryption & STUN 11. Voice Quality Silence Suppression, Packets Loss Management 12. DTMF - (Dual Tone

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    ...Protocol) & RTCP (RTP Control Protocol) 8. Transport Mode TCP, User Datagram Protocol (UDP), TCP to UDP Interworking 9. Codecs G.723, G.729, AMR (Adaptive Multi-Rate Compression), iLBC, G.711, GSM. 10. Network Address Translation (NAT) Traversal Through Encryption & STUN 11. Voice Quality Silence Suppression, Packets Loss Management 12. DTMF - (Dual Tone

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    ...contacts from the phone. 3. synchronize phone contacts to server. 4. Time, call status logs, last calls log and Redial 5. STUN 6. Support for codecs OPUS, g729, GSM, g711 and iLBC. 7. Built-in tunnel for passing through voip blockades 8. Enhanced battery usage (should not kill the battery) a. The source code should fully documented. b. Doesn't need

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    SIP Dialer Udløbet left

    ...Protocol) & RTCP (RTP Control Protocol) 8. Transport Mode TCP, User Datagram Protocol (UDP), TCP to UDP Interworking [log ind for at se URL] G.729, AMR (Adaptive Multi-Rate Compression), iLBC, G.711, GSM. [log ind for at se URL] Address Translation (NAT) Traversal Through Encryption & STUN [log ind for at se URL] Quality Silence Suppression, Packets Loss Management 12...

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    ...as below    Technical Requirements: The application should support the following codecs: G.722.1, G.723.1, G.726, G.728, G.711, G.729, G.723.1, iLBC,: G.711 (A and m-law),G.729A/B/D/E, AMR, GSM 6.10/EFR, iLBC, Speex · Voice: Real time Quality Monitoring (MOS) 3 independent phone lines Auto-Answer/Do Not Disturb Call Forwarding Full Duplex Audio Recording

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    ...as below    Technical Requirements: The application should support the following codecs: G.722.1, G.723.1, G.726, G.728, G.711, G.729, G.723.1, iLBC,: G.711 (A and m-law),G.729A/B/D/E, AMR, GSM 6.10/EFR, iLBC, Speex · Voice: Real time Quality Monitoring (MOS) 3 independent phone lines Auto-Answer/Do Not Disturb Call Forwarding Full Duplex Audio Recording

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    We have a SIP based mobile application for iOS and Android. It uses all standard codecs such as G729, G711, G722, Opus, iLBC - If we use G729, it uses 45+ kbps bandwidth. - We are looking for a module that can compress the SIP signals and RTP packets by upto 60% - We are open to either compression or using any other codec that is optimized on

    €879 (Avg Bid)
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    mobile dialer Udløbet left

    Project Description: Description Description 1) User friendly interface 2) Dialer supports SIP for signaling. 3) Dialer supports G729, G.711, G723.1, iLBC codec for sending audio & Video data. 4) Dialer should be able to run behind NAT or on private IP. 5) client can see the remaining balance in real time. 6) Dialer should be fully customizable

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    ...Protocol) & RTCP (RTP Control Protocol) 8. Transport Mode TCP, User Datagram Protocol (UDP), TCP to UDP Interworking 9. Codecs G.723, G.729, AMR (Adaptive Multi-Rate Compression), iLBC, G.711, GSM. 10. Network Address Translation (NAT) Traversal Through Encryption & STUN 11. Voice Quality Silence Suppression, Packets Loss Management 12. DTMF - (Dual Tone

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    ...Protocol) & RTCP (RTP Control Protocol) 8. Transport Mode TCP, User Datagram Protocol (UDP), TCP to UDP Interworking 9. Codecs G.723, G.729, AMR (Adaptive Multi-Rate Compression), iLBC, G.711, GSM. 10. Network Address Translation (NAT) Traversal Through Encryption & STUN 11. Voice Quality Silence Suppression, Packets Loss Management 12. DTMF - (Dual Tone

    €5160 (Avg Bid)
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    mobile dialer Udløbet left

    Project Description: Description Description 1) User friendly interface 2) Dialer supports SIP for signaling. 3) Dialer supports G729, G.711, G723.1, iLBC codec for sending audio & Video data. 4) Dialer should be able to run behind NAT or on private IP. 5) client can see the remaining balance in real time. 6) Dialer should be fully customizable

    €901 (Avg Bid)
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    ...Protocol) & RTCP (RTP Control Protocol) 8. Transport Mode TCP, User Datagram Protocol (UDP), TCP to UDP Interworking 9. Codecs G.723, G.729, AMR (Adaptive Multi-Rate Compression), iLBC, G.711, GSM. 10. Network Address Translation (NAT) Traversal Through Encryption & STUN 11. Voice Quality Silence Suppression, Packets Loss Management 12. DTMF - (Dual Tone

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    ...& RTCP (RTP Control Protocol) 8. Transport Mode TCP, User Datagram Protocol (UDP), TCP to UDP Interworking 9. Codecs G.723, G.729, AMR (Adaptive Multi-Rate Compression), iLBC, G.711, GSM. 10. Network Address Translation (NAT) Traversal Through Encryption & STUN 11. Voice Quality Silence Suppression, Packets Loss Management 12. DTMF - (Dual Tone

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    ...prototying)is done through the Lua [log ind for at se URL] mirrors the c++ namespce as closely as possible strongswan strongswan is a complete IPsec implentation for the Linux,Andriod,Maemo,FreeBSD,and MAC OSX operating [log ind for at se URL] interoperates with most other IPSEC-based VPN product via the IKEV2 Or IKEV1 key exchange protocals A rich choice of moduler

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    Hi, I need someone who can help me convert iLBC encoded .caf files to MP3, this needs to occur on an Linux instance of EC2. Please contact me with questions. Best, Don ## Deliverables here is a sample file- if you can covert this to mp3 please talk to me asap! Don

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    Hello I need the Linphone Web products asap with all its codec. Must Required : - G729, AMR, ILBC, SILK Codecs - MPEG4 and H264 Please read it first and Reply me after [log ind for at se URL] want ready made solution not want to new develop Thanks

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    ...Secure communications : zRTP (available only under GNU GPL v2 license), TLS, SRTP Advanced features : Audio codecs : G711, SPEEX, G722, AMR-WB (G722.2), GSM 6.10, AMR-NB, ILBC, SILK, G729 Integration with push notification (requires compatible SIP server) Nat friendly: guesses NAT address for SIP messages, uses STUN for RTP streams. Integrated ICE

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    SIP Dialer Udløbet left

    ...Protocol) & RTCP (RTP Control Protocol) 8. Transport Mode TCP, User Datagram Protocol (UDP), TCP to UDP Interworking [log ind for at se URL] G.729, AMR (Adaptive Multi-Rate Compression), iLBC, G.711, GSM. [log ind for at se URL] Address Translation (NAT) Traversal Through Encryption & STUN [log ind for at se URL] Quality Silence Suppression, Packets Loss Management 12...

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    ... Compliant with IETF standards SIP and XMPP Compliant with IETF, 3GPP, and CableLabs standards STUN, TURN, and ICE Plug-and-play voice codecs including G.711, G.729, GSM, iLBC, SirenTM, Speex, and more Voice processing (AEC, AGC, CNG, NS, VAD) for hands-free handsets and tablets Conferencing for up to 8 parties Peer-to-peer media transport

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    ... Compliant with IETF standards SIP and XMPP Compliant with IETF, 3GPP, and CableLabs standards STUN, TURN, and ICE Plug-and-play voice codecs including G.711, G.729, GSM, iLBC, SirenTM, Speex, and more Voice processing (AEC, AGC, CNG, NS, VAD) for hands-free handsets and tablets Conferencing for up to 8 parties Peer-to-peer media transport

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    ...& RTCP (RTP Control Protocol) 8. Transport Mode TCP, User Datagram Protocol (UDP), TCP to UDP Interworking 9. Codecs G.723, G.729, AMR (Adaptive Multi-Rate Compression), iLBC, G.711, GSM. 10. Network Address Translation (NAT) Traversal Through Encryption & STUN 11. Voice Quality Silence Suppression, Packets Loss Management 12. DTMF - (Dual Tone

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    ... Compliant with IETF standards SIP and XMPP Compliant with IETF, 3GPP, and CableLabs standards STUN, TURN, and ICE Plug-and-play voice codecs including G.711, G.729, GSM, iLBC, SirenTM, Speex, and more Voice processing (AEC, AGC, CNG, NS, VAD) for hands-free handsets and tablets Conferencing for up to 8 parties Peer-to-peer media transport

    €660 - €1320
    Forseglet
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    ... Compliant with IETF standards SIP and XMPP Compliant with IETF, 3GPP, and CableLabs standards STUN, TURN, and ICE Plug-and-play voice codecs including G.711, G.729, GSM, iLBC, SirenTM, Speex, and more Voice processing (AEC, AGC, CNG, NS, VAD) for hands-free handsets and tablets Conferencing for up to 8 parties Peer-to-peer media transport

    €660 - €1320
    Forseglet
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    ... Compliant with IETF standards SIP and XMPP Compliant with IETF, 3GPP, and CableLabs standards STUN, TURN, and ICE Plug-and-play voice codecs including G.711, G.729, GSM, iLBC, SirenTM, Speex, and more Voice processing (AEC, AGC, CNG, NS, VAD) for hands-free handsets and tablets Conferencing for up to 8 parties Peer-to-peer media transport

    €660 - €1320
    Forseglet
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    Call Egypt Udløbet left

    ...Protocol) & RTCP (RTP Control Protocol) 8. Transport Mode TCP, User Datagram Protocol (UDP), TCP to UDP Interworking 9. Codecs G.723, G.729, AMR (Adaptive Multi-Rate Compression), iLBC, G.711, GSM. 10. Network Address Translation (NAT) Traversal Through Encryption & STUN 11. Voice Quality Silence Suppression, Packets Loss Management 12. DTMF - (Dual Tone

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