SIPDomainCalling
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GNU SIP Witch is a service for the SIP protocol that is analogous to what Jabber does to XMPP, that is it offers registration, routing, and messaging between arbitrary SIP endpoints and resources. GNU SIP Witch was selected as part of the FSF initiative to replace Skype with free software, however there are additional oppertunities that are very relevant to Ubuntu. SIP Witch can also be used as a local backend to unifiy the telephony experience of desktop users in Telepathy and other desktop SIP user agents by automatically mediating access to local telephony resources such as SIP devices, desktop modems, gsm radio, idsn adapters, or local telephony servers, as well as remote VoIP service providers and enabling direct and secure call peering over the public Internet. The goal is to identify the work required to implement a flexible model allowing for an effective telephony experience for Ubuntu users. | GNU SIP Witch is a service for the SIP protocol that is analogous to what Jabber does to XMPP, that is it offers registration, routing, and messaging between arbitrary SIP endpoints and resources. GNU SIP Witch was selected as part of the FSF initiative to replace Skype with free software, however there are additional opportunities that are very relevant to Ubuntu. SIP Witch can also be used as a local back-end to unify the telephony experience of desktop users in Telepathy and other desktop SIP user agents by automatically mediating access to local telephony resources such as SIP devices, desktop modems, gsm radio, idsn adapters, or local telephony servers, as well as remote VoIP service providers and enabling direct and secure call peering over the public Internet. The goal is to identify the work required to implement a flexible model allowing for an effective telephony experience for Ubuntu users. |
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Destkop domain calling services will empower users to unify their telephony experience in new ways on Ubuntu. This includes creating and participating in secure calling networks over the Public Internet, as well as accessing both local and remote telephony resources and services in a unified desktop client experience. | Desktop domain calling services will empower users to unify their telephony experience in new ways on Ubuntu. This includes creating and participating in secure calling networks over the Public Internet, as well as enabling access to both local and remote telephony resources and services in a unified client experience. |
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There is a need for an Internet scalable solution for VoIP that offers intercept-free communiation and which does not rely upon the control of a single provider or the use of non-standard's compliant clients. | There is need to be able to use any resource that is available for communicating with people, whether a local modem device, an existing service provider, a private IP-PBX, or even directly in a secure manner using the public Internet, and to be able to do so automatically without requiring user intervention when situations such as connectivity to external services change. To enable a complete telephony experience we can begin with unifying the client experience such as with Telepathy. For example, we want to begin with one universal contact for telephony be it a phone number or a sip uri, and a consistent manner for a single client to connect to said user with a single contact without requiring our users to determine where or how the call is actually placed, what service entry to use, etc. The user experience should be seamless, and hence the purpose of proposing a local mediating service for the SIP protocol. |
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User wants to call other users at remote locations. For this, we will use URI dialing, which means SIP witch will map email-like id's into local extensions. |
User wants to call other users at remote locations. This might be by SIP uri or by a telephone number. Whichever use, it should not matter for the user to have to think about which profile or account to "use" and hence should be handled consistently from the client perspective regardless if the call is ultimately handed off for completion using a modem device, gsm radio, some SIP service provider, a local or remote IP-PBX, or the public Internet directly for secure URI peer to peer calling. Nor should the user need to consider which set of SIP settings for NAT issues are required based on who the user is calling and how the call is being completed. |
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For Lucid+1: * Introduction of mysql or sqlite plugin for keeping user extensions in a database * xmlrpc backend for managing configuration and db * new front-end app with systray for remote status monitoring of a sipwitch server |
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* Gateways. This spec offers a way to unify client communication with IP-PBX services offering pstn gateways, that may also run on Ubuntu, but it does not define or address these. | |
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== Action Items == | == BoF agenda and discussion == From UDS Lucid: "or why the desktop telephony user experience has been miserable so far" History: * Iphone in 1995 was offered by an Israeli company, worked on dialup * Gnomemeeting / ekiga / empathy * Secure voice call a few years back, but required knowledge of each party's IP FSF is trying to replace skype with a Free software alternative (sip witch): http://www.fsf.org/campaigns/priority.html#skypereplacement Existing software: * SFLphone * Packaged, third-party, GPL, not for Karmic yet * GizmoProject / Gizmo5 http://www.gizmo5.com - not available anymore, non-free, acquired by Google * Qutecom (ex wengophone) * Twinkle * Empathy / telepathy * Ekiga Empathy is our default app for communication, we should figure out how to make that experience good Usability issues: * Too many options exposed in the UI * NAT problems * Even more problems when using multiple interfaces (vpn) * Google talk: enter username and password, and it just works * Have configurations for multiple providers, as a drop-file just as we do with broadband providers SIP Witch: SIP Witch will take care of encryption and NAT and finding an available service. So we tell telepathy to just talk to SIP Witch. * Integrate it to sofia-sip-bin / sophia-sip-telepathy ? http://savannah.gnu.org/projects/sipwitch Integration into Ubuntu One? Issues: * Create desktop telephony use case that make sense * Current telepany does not provide secure calls Action Items: * Make sure sip witch is packaged in Ubuntu * SIP Witch needs some improvements * Easier way to add provider information into sip witch * Document current SIP providers and how to use them with SIP Witch * Add logic to abstract using SIP providers (and help them add their config info) * Password synchronization (betwixt local user and service) * Work with more services * Make it easier for people to test, get lots of feedback/bugs * Add link to Answers in Launchpad so problems/info are *asked* first then * Add some documentation to the wiki, advertise its existence * Link to Freenode webchat for easy " I want to test " pairing For Lucid+1?: * Add logic to telepathy to recognize sip witch to automatically configure * Integrated with UbuntuOne(?) |
Launchpad Entry: desktop-lucid-sip-domain-calling
Created: 2009-10-10
Contributors: DavidSugar
Packages affected: sipwitch
Summary
GNU SIP Witch is a service for the SIP protocol that is analogous to what Jabber does to XMPP, that is it offers registration, routing, and messaging between arbitrary SIP endpoints and resources. GNU SIP Witch was selected as part of the FSF initiative to replace Skype with free software, however there are additional opportunities that are very relevant to Ubuntu. SIP Witch can also be used as a local back-end to unify the telephony experience of desktop users in Telepathy and other desktop SIP user agents by automatically mediating access to local telephony resources such as SIP devices, desktop modems, gsm radio, idsn adapters, or local telephony servers, as well as remote VoIP service providers and enabling direct and secure call peering over the public Internet. The goal is to identify the work required to implement a flexible model allowing for an effective telephony experience for Ubuntu users.
Release Note
Desktop domain calling services will empower users to unify their telephony experience in new ways on Ubuntu. This includes creating and participating in secure calling networks over the Public Internet, as well as enabling access to both local and remote telephony resources and services in a unified client experience.
Rationale
There is need to be able to use any resource that is available for communicating with people, whether a local modem device, an existing service provider, a private IP-PBX, or even directly in a secure manner using the public Internet, and to be able to do so automatically without requiring user intervention when situations such as connectivity to external services change.
To enable a complete telephony experience we can begin with unifying the client experience such as with Telepathy. For example, we want to begin with one universal contact for telephony be it a phone number or a sip uri, and a consistent manner for a single client to connect to said user with a single contact without requiring our users to determine where or how the call is actually placed, what service entry to use, etc. The user experience should be seamless, and hence the purpose of proposing a local mediating service for the SIP protocol.
User stories
User wants to run a simple phone system for several softphones they have at home, and dial them locally as extensions and have basic pbx features like call forwarding, multi-phone ring groups, etc.
User wants to call other users at remote locations. This might be by SIP uri or by a telephone number. Whichever use, it should not matter for the user to have to think about which profile or account to "use" and hence should be handled consistently from the client perspective regardless if the call is ultimately handed off for completion using a modem device, gsm radio, some SIP service provider, a local or remote IP-PBX, or the public Internet directly for secure URI peer to peer calling. Nor should the user need to consider which set of SIP settings for NAT issues are required based on who the user is calling and how the call is being completed.
Design
We will be making use of existing SIP compliant softphones, many of which have already been packaged for Ubuntu. SIP Witch simply coordinates these devices in a local dialing plan, and offers uri routing that is resolved through DNS for calling remote users.
Since SIP Witch only mitigates SIP and will soon offer media packet forward RTP for SIP devices behind a NAT, while still establishing direct peer-to-peer communication between endpoints, it has very little overhead and no issues with patent encumbered codecs. Because peer to peer media connections are used between endpoints, sipwitch can operate directly with, manage, and scale "Social Key Verification" systems such as ZRTP. Since all users can use direct URI dialing to contact users at other locations there is no need for a central "service provider" or directory.
The primary feature is being able to run a SIP Witch "service" on Ubuntu, and this is a packaging issue alone. Exposing this service to management, that is, making it possible for ordinary human beings to in some even basic manner configure a SIP Witch installation for this specific mission at minimum without hand editing config files, and the completion of packet forward integration in the daemon itself, is the only Ubuntu-specific feature proposed for Lucid.
Implementation
For Lucid:
- Packaging of sipwitch daemon (nearly done)
- Completion of packet forward feature for proper NAT support
- Simple gui app (maybe done in quickly) to generate a configuration file
Ability to select gui config app from System->Administration
- Basic documentation for this
- Cache user sip hashes when user passwords change via pam stack and new /etc/sipwitch.db
Effort
Lucid:
- Packet forwarding may take a few days to complete and test
- new digest cache a few days
- gui app time unknown
- Packaging should be about a day
- Total effort ~1-2 weeks
UI Changes
- New gui app for system-admin to create sipwitch configurations
Code Changes
Migration
Test/Demo Plan
Unresolved issues
- Gateways. This spec offers a way to unify client communication with IP-PBX services offering pstn gateways, that may also run on Ubuntu, but it does not define or address these.
BoF agenda and discussion
From UDS Lucid:
"or why the desktop telephony user experience has been miserable so far"
History:
- Iphone in 1995 was offered by an Israeli company, worked on dialup
- Gnomemeeting / ekiga / empathy
- Secure voice call a few years back, but required knowledge of each party's IP
FSF is trying to replace skype with a Free software alternative (sip witch): http://www.fsf.org/campaigns/priority.html#skypereplacement
Existing software:
- SFLphone
- Packaged, third-party, GPL, not for Karmic yet
GizmoProject / Gizmo5 http://www.gizmo5.com - not available anymore, non-free, acquired by Google
- Qutecom (ex wengophone)
- Twinkle
- Empathy / telepathy
- Ekiga
Empathy is our default app for communication, we should figure out how to make that experience good
Usability issues:
- Too many options exposed in the UI
- NAT problems
- Even more problems when using multiple interfaces (vpn)
- Google talk: enter username and password, and it just works
- Have configurations for multiple providers, as a drop-file just as we do with broadband providers
SIP Witch: SIP Witch will take care of encryption and NAT and finding an available service. So we tell telepathy to just talk to SIP Witch.
- Integrate it to sofia-sip-bin / sophia-sip-telepathy ?
http://savannah.gnu.org/projects/sipwitch
Integration into Ubuntu One?
Issues:
- Create desktop telephony use case that make sense
- Current telepany does not provide secure calls
Action Items:
- Make sure sip witch is packaged in Ubuntu
- SIP Witch needs some improvements
- Easier way to add provider information into sip witch
- Document current SIP providers and how to use them with SIP Witch
- Add logic to abstract using SIP providers (and help them add their config info)
- Password synchronization (betwixt local user and service)
- Work with more services
- Make it easier for people to test, get lots of feedback/bugs
- Add link to Answers in Launchpad so problems/info are *asked* first then
- Add some documentation to the wiki, advertise its existence
- Link to Freenode webchat for easy " I want to test " pairing
For Lucid+1?:
- Add logic to telepathy to recognize sip witch to automatically configure
Integrated with UbuntuOne(?)
Specs/SIPDomainCalling (last edited 2009-11-30 20:38:15 by pool-72-75-154-242)