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SIP and RTSP for Unified Voicemail

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0% found this document useful (0 votes)
18 views17 pages

SIP and RTSP for Unified Voicemail

Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as PPT, PDF, TXT or read online on Scribd

Kundan Singh and

Henning Schulzrinne, Columbia University


Agenda
• Introduction
• Requirements
• Architecture
• Issues
• Implementation
• Q/A
Traditional voice mail system
Dial 853-8119

Phone is ringing
Alice Bob
939-7063 .. The person is not available now 853-8119
please leave a message ...
... Your voice message ...

Disconnect

Bob can listen to his voice mails by dialing some number.


Problems
• Voice mail system tied to PBX or phone
company (if CFB)
• Integration of video, fax, whiteboard?
• How to integrate with Internet telephony?
• How to integrate with email, web and other
user applications?
Existing solutions
• Voice Profile for Internet Messaging
(VPIM)
• Web-based unified messaging systems with
personalized PSTN voice mail number.
Design Goals
• Message recording and playback
• Universal access: web, email, VoIP, PSTN
• Email notification
• Scalability for large domains
• Separable from ITSP or ISP
• Reuse existing infrastructure
• Media-agnostic
• Tool-agnostic
• Telephony interface - DTMF
Why SIP and RTSP ?
• Use SIP for accepting voice/video calls (other
services, and infrastructure for Internet telephony)
• RTSP for storage and access of voice messages.
• RTSP already in common use, e.g., RealPlayer
• Large-scale RTSP MoD servers exist.
• Easy integration with web, email, video and fax.
• Access from PSTN using a gateway.
Architecture Bob

INVITE bob@[Link]
[Link]
INVITE
bob@[Link]

REGISTER bob@[Link]
Alice

INVITE bob@[Link]

[Link]

The voice mail server registers with the SIP proxy, sipd, on behalf of every user.
Alice calls bob@[Link] through SIP proxy.

SIP proxy forks the request to Bob’s phone as well as to a voicemail server.
Architecture Bob

[Link];

CANCEL

200 OK
Alice

200 OK

RTP/RTCP v-mail
[Link];
After 10 seconds vm contacts the
RTSP server for recording. SETUP
vm accepts the call.
Sipd cancels the other branch and ...
rtspd
...accepts the call from Alice.
Now user message gets recorded
Architecture Bob

[Link]

Email to bob@[Link]
BYE
Alice
RTP
200 OK

v-mail
[Link]

Once the call is closed by Alice, vm sends an


email to Bob informing him of the arrival of a voice mail.
Architecture Bob

[Link]

INVITE bob-526156-retrieve@[Link]
Alice
RTP

Quick-time

v-mail
[Link]
RTSP
Bob can listen to voice mail using either SETUP/PLAY
an RTSP client like QuickTime or ...
...by calling the v-mail using SIP. rtspd
…or by visiting his web-based voicemail account
Architecture
Alternatives
– The SIP phone redirects the call to voice mail
after 10 seconds.
– The SIP proxy is configured to forward the call
to voice mail if your phone is busy or there is
no response (static, or using sip-cgi, CPL)
– Voice mail server acts as another phone for the
user but delays accepting the call by 10
seconds, with CANCEL if user picks up.
Issues
• Call reclaiming
• Deleting voice/video mail
• Integration with PSTN phone
• Integration with VPIM, IMAP, POP3
Implementation

• Prototype
system.
• Recording
and playback
using .au
files.

[Link]
Implementation
Features

• Integration with web/email for more control


over voicemail configuration (e.g., folder
management, email notification.)
• Web based voice mail accounts for users
(Similar to Hotmail)

• Retrieval using RTSP clients (Quicktime), SIP


user agent (e*phone) or Web browser.
Implementation
Future

• DTMF based navigation


• Support for other media formats in rtspd
• Deployment
• Multimedia mail SIP retrieval
Conclusion
• SIP and RTSP - good framework for unified
messaging
• Integration of voice/video mail/answering
machine, email, instant messaging, fax, etc.
Conclusion
Wide range of applicability

Campus/corporate network

rtspd
Internet
sipum sipum

rtspd

External application
Within a domain
service provider

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