Find Jobs
Hire Freelancers

SIP (Session Initiation Protocol) UA

$30-5000 USD

Inställt
Publicerad över 17 år sedan

$30-5000 USD

Betalning vid leverans
SIP User Agent Requirements 1. MUST use freebie SIP UA available for commercial use (e.g reciprocate or sipx) 2. MUST use .NET or provide COM interface. Language must be C# or C++. 3. Features 1. MUST comply with RFC 3264 for SIP and associated RFCs as required. 2. MUST support authentication using userid/password (API must be provided). But the authentication should be optional for the UA to function. MUST provide options inside code to challenge INVITE, BYE, REGISTER. 3. MUST support registration (API must be provided). But Registration should be optional for the UA to function. 4. MUST support making, receiving calls (voice only). i. Codecs: G.711, G.723, G.729, iLBC, Speex 1. MUST support call transfer, hold 2. MUST support ending active calls and call in progress. API must be provided. 3. MUST support calling to SIP URIs and TEL URIs. 4. OPTIONAL (for extra $$) RTP media (play from network as well as send to network) must be supported. A freebie RTP library may be used. 5. MUST provide call back functions for each call state change (e.g ringing, connected, disconnected etc). ## Deliverables 1) Complete and fully-functional working program(s) in executable form as well as complete source code of all work done. 2) Deliverables must be in ready-to-run condition, as follows (depending on the nature of the deliverables): a) For web sites or other server-side deliverables intended to only ever exist in one place in the Buyer's environment--Deliverables must be installed by the Seller in ready-to-run condition in the Buyer's environment. b) For all others including desktop software or software the buyer intends to distribute: A software installation package that will install the software in ready-to-run condition on the platform(s) specified in this bid request. 3) All deliverables will be considered "work made for hire" under U.S. Copyright law. Buyer will receive exclusive and complete copyrights to all work purchased. (No GPL, GNU, 3rd party components, etc. unless all copyright ramifications are explained AND AGREED TO by the buyer on the site per the coder's Seller Legal Agreement). * * *This broadcast message was sent to all bidders on Thursday Dec 28, 2006 2:27:41 PM: I am only looking for APIs and module which may require its own configuration (as a re-readable XML file containing URI domain, codecs etc or using APIs itself). Configuration should be changeable at runtime. For the codec negotiation, multiple codecs are allowed, so the API must allow configuration of multiple codecs which can be sent in the INVITE. Codec order should also be provisionable which is used to list codecs in offer and accept in response. There is no GUI required for this project. Our application will contain the GUI. You may provide your own GUI for testing etc. * * *This broadcast message was sent to all bidders on Tuesday Jan 2, 2007 8:52:56 AM: Updated Requirements: SIP User Agent Requirements 1. MUST use reciprocate SIP stack. 2. MUST use .NET or provide COM interface. Language must be C# or C++. 3. Features a. MUST comply with RFC 3264 for SIP and associated RFCs as required. b. MUST support authentication using userid/password (API must be provided). But the authentication should be optional for the UA to function. MUST provide options inside code to challenge INVITE, BYE, REGISTER. c. MUST support registration (API must be provided). But Registration should be optional for the UA to function. d. MUST support making, receiving calls (voice only). i. Codecs: G.711, G.723, G.729, iLBC, Speex e. MUST support call transfer, hold f. MUST support ending active calls and call in progress. API must be provided. g. MUST support calling to SIP URIs and TEL URIs. h. MUST provide RTP media (play from network as well as send to network) must be supported. A freebie RTP library MUST be used (like sipXmediaLib or Vovida RTP, JRTPLIB or ortp from Linphone). Full RTP and sound plumbing must be provided. i. MUST provide call back functions for each call state change (e.g ringing, connected, disconnected etc). Additions/Modifications/More details provided on Jan 1, 2007: 4. MUST be tested to work with with SER (SIP Express Router [login to view URL], Asterisk [login to view URL] and the Pingtel Interop site [login to view URL]) 5. 1, has been changed to use reSIProcate as a MUST. This makes the project results predictable for us. 6. 3(h) which was previously optional has been now made a MUST since we know that the software cannot properly be tested if this feature is not there. 7. MUST provide the necessary multi-thread safe code and API calls for application to easily launch and “forget?? about SIP code. We understand that the calling application must launch some of the vital threads. The application layer needs to be as simple as possible. 8. RTP Playback and Ringing devices should be separately controllable through API calls. User level code MUST not be used to play RTP. RTP packets received from network should directly be played from kernel level to sound device without passing them onto application or usermode. Same holds for packets sent to the network. They should not unnecessarily pass through the user level. If this is not possible due to some issues, it should be discussed with us. 9. DTMF event detection and generation is a must. Conference bridge mixing, and all other features offered by the sipXmediaLib are expected as well in the UA. Correct choice of the RTP software will make this zero effort. 10. SIP over UDP/TCP and TLS must be supported completely. DTLS, SCTP should be within future scope. Again this is zero effort if reciprocate is chosen, but UA should not undermine this. IMPORTANT NOTE: Since this software is developed over a lot of open source software already existing, the bid must take that into account. The project is about integration of these and providing APIs on top. It is up to us what we decide to do with the code. We may use it in our application, or decide to give it back to the open source community at which point the code and the names of developers may be made public (unless we are asked not to do that). It is important for the bidders to understand the impact of this on desirable code quality (high for open source software created in this project and norms should be followed while coding) and the future opportunities it may create for the bidder (since we will get out of the middle if someone else also approaches you for enhancements on the open source UA created. Bidders should take this into account while bidding). * * *This broadcast message was sent to all bidders on Friday Jan 5, 2007 7:58:36 AM: We have cleaned up the requirements document to make it simpler and not restate what is already supported by reSIProcate/sipxmedialib. The timeframes for this project are 1 month. We also have received information from SIPX and reSIProcate developers that there have been more than three occasions that the sipxmedialib and reSIProcate have been put together easily. So the project feasibility is already there. See below for requirements. SIP User Agent Requirements (aka Softphone APIs w/o GUI) 1. MUST use reciprocate SIP stack for signaling and sipxmedialib for media integration on the Windows XP/2000 environment. 2. MUST use .NET or provide COM interface to our application. Language must be C# or C++. 3. Features a. API: API must be presented as an object interface since reciprocate is object oriented code. The following APIs are required: i. Register ii. Make, Hangu p, Cancel Calls iii. Call Transfer (Blind and Attended) iv. Hold, Mute v. Authentication (challenge and response) for Registration, making calls, hanging up a call. vi. Must be able to handle DTMF events in RTP and SDP vii. APIs must support all URIs supported by the reciprocate stack (SIP, SIPS, TEL). viii. Make additional calls if current call is on hold. Manage multiple ongoing calls simultaneously (up to 10 calls say). ix. Create a Conference and mix media (using sipxmedialib). Add existing call to conference. Create a new call in conference. Add an incoming call into a conference. x. An initialization routine which contains code for initializing the reciprocate stack and launches all necessary threads. xi. API to read config file (see below), make changes to it and read off specific configuration. xii. call back API for each call state change (e.g ringing, connected, disconnected etc). b. Configuration File (XML): This file must be re-readable at run-time and any changes done must take effect w/o restarting the API (ongoing calls/activity must not be disturbed) i. Must contain codecs which will be sent in SDP offer or which will be compared with incoming offer to signal media layer of selected codec. Codec priority should be specifiable as well. ii. Registration (enabled or not) iii. Destination Server URIs (DNS Address, Port, Transport Protocol) and priorities which specify the order in which the servers will be tried in case the call fails to connect through one URI. A regular expression should also be provided for each Server URI which must yield a positive match for the server to be used. iv. STUN Server address (reciprocate supports NAT Traversal using STUN) v. Call Forwarding On/Off. If On, the new URI must be provisionable. 4. Media a. Codecs: G.711, G.723, G.729, iLBC, Speex may be supported in SDP and configuration file. G.711 must be integrated in as i t is already part of the sipxmedialib. b. RTP processing should be completely within kernel using the sipXmediaLib. c. Input, Output and Ringing devices should be separately controllable through API. 5. Testing a. MUST be tested to work with with SER (SIP Express Router [login to view URL], Asterisk [login to view URL] and the Pingtel Interop site [login to view URL]) IMPORTANT NOTE: All rights, IP and source code will be owned by us. No copyright notices or author names should be put into any source files or other files. Any third party source code may be used, provided permission is obtained from us. Website for obtaining reciprocate code: [login to view URL] Websites for obtaining sipxmedialib code: Please, check out sipXtapi branch with following command: svn checkout [login to view URL] . You may also try sipXtapi-media-update branch: svn checkout http://s [login to view URL] . [login to view URL] ## Platform Windows XP/2000
Project ID: 3968967

Om projektet

1 anbud
Distansprojekt
Senaste aktivitet 17 år sedan

Ute efter att tjäna lite pengar?

Fördelar med att lägga anbud hos Freelancer

Ange budget och tidsram
Få betalt för ditt arbete
Beskriv ditt förslag
Det är gratis att registrera sig och att lägga anbud på uppdrag
1 frilansar lägger i genomsnitt anbud på $1 615 USD för detta uppdrag
Använd avatar
See private message.
$1 615 USD Om 14 dagar
4,7 (10 omdömen)
6,4
6,4

Om kunden

Flagga för UNITED STATES
United States
4,8
7
Medlem sedan juni 11, 2006

Kundverifikation

Tack! Vi har skickat en länk för aktivering av gratis kredit.
Något gick fel med ditt e-postmeddelande. Vänligen försök igen.
Registrerade Användare Totalt antal jobb publicerade
Freelancer ® is a registered Trademark of Freelancer Technology Pty Limited (ACN 142 189 759)
Copyright © 2024 Freelancer Technology Pty Limited (ACN 142 189 759)
Laddar förhandsgranskning
Tillstånd beviljat för geolokalisering.
Din inloggningssession har löpt ut och du har blivit utloggad. Logga in igen.