Switzernet
PSE, Bâtiment A, EPFL
1015 Lausanne
Student projects
Switzernet
2008-04-03
[Updated on 2008-08-28]
[Updated on 2008-09-04]
[provider_rates] An application to compare different provider rates and optimize our routing table
[customer_db] customer database analyses
[referral] analysis of referrals
[support] customer support analyses
[signalling] call setup tests based on signalling
[voice_analysis] call setup tests based on voice analysis
[online_monitoring] online monitoring
[radius_asterisk] RADIUS protocol integration in Asterisk
[provider_rates] An application to compare different provider rates and optimize our routing table
Every month, we receive updated price lists from our different telephony providers. From this information, our routing table (the priority to which we send traffic to a given operator depending on the destination country) changes, based on different criteria (quality, cost, etc.).
The goal of this project is to develop an application to compare the rates of all our providers, and optimize our routing table accordingly. Data will be extracted from the price lists, and stored in a database for fast queries. The results must be accessible through a web interface.
Some of the key points:
- Prefixes sent by different operators will differ. They can use a finer subdivision for some countries, or only propose certain destinations. The first task is to find a unified subset of prefixes to allow comparison between operators.
- Upload/download of price lists must be easy, adapting to different data templates (Excel or CSV files).
- Rates comparison must take into account our volume of calls (monthly minutes) to each destination.
- The given prices must be transformed (currency conversion, consider taxes, rounding, connection fees, peak/off-peak hours, etc.) to a standard format.
- The current routing priorities for each destination must also be shown. This will be queried from our running database.
Additional features, depending on time:
- Finer criteria for comparison (voice quality, average call success rate, etc.)
- Direct import of the optimized routing table in our billing engine.
- Graphs showing the evolution of the rates for a given destination.
- Suggested technologies: PHP, MySQL, Perl, ...
[retail_rates] comparison of prices and of billing schemes with other operators for selected destinations
- you can select a group of international and/or national destinations
- you can select a group of providers: VOIP, traditional telephony (such as Swisscom, Sunrise, Tele2), national, or international (such as Skype)
- call setup fee, rounding of call duration, etc
- call setup and rounding’s impact on price depends on destination (ACD, etc)
- the impact of the rounding on the price (statistical computation on a number of calls)
- list of references is required
References:
http://switzernet.com/public/071023-rates-to-swiss-mobiles/
http://switzernet.com/public/070626-skype-rounding/
http://switzernet.com/public/070626-rounding-of-minutes/
http://switzernet.com/public/070625-rounding-and-callsetup/
http://switzernet.com/public/070619-comparis-swiss-mobiles/
http://switzernet.com/public/070625-comparis-voip-ch-de-fr-us-uk/
[customer_db] customer database analyses
- statistics on contracts
- type of signature: post, scan, e-signature, fax
- date of signature
- date of request of cancellation
- date of termination of the contract
- time diagram between the online inscription and signed confirmation from customer
- customer’s profile analyses (nationality, age)
- most called destinations per customer
- customer’s location analyses
- sales statistics
- per reseller, location, etc
[referral] analysis of referrals
- define a customer stamp for keeping trace of referrals [ch1], [ch2]. The referral information can be retrieved from subscription mails (contracts@) and entered to billing notepads [ch1], [ch2]
- create a graph of referrals and analyze the graph (who brought whom, who in its turn brought how many, etc).
[support] customer support analyses
- statistics on support@ and billing@ replies depending on the class and type of the answer
Reference: support-emails
Chart 1. Monthly statistics of support@ replies merged with the statistics of billing@ (as of 2008-09-01). Statistics are done depending on class only.
[signalling] call setup tests based on signalling
- ASR: answer seizure ratio (for example you can make 20 calls to a given destination and report the number of connections and the number of failed attempts)
- PDD: post dial delay (the average time in seconds from the moment you dialled the last digit and pressed the send button until the moment the target phone starts ringing)
- make measurements and provide comparisons with classical telephone lines
- the parameters strongly depend on the type of the destination (mobiles or landlines will exhibit different parameters)
- computer software can be developed, which calls given destinations regularly and create appropriate charts
[voice_analysis] call setup tests based on voice analysis
- The requested software shall connect at regular intervals to all of our telephony SIP servers and shall make test calls to a given PSTN number (through each of SIP servers). The test phone number, once connected, plays a standard voice message.
- The software must record the message and compare it with its original
- The software must only figure out if the recorded message corresponds to the original (no other analysis). The comparison must say ‘matched’ or ‘not matched’. Based on the voice comparison, the program must determine if the call attempt was successful or not.
- The test calls must be placed every 2 minutes, hourly statistics should be generated (success rate, failure rate, average connection time, etc) per SIP server.
- Students will have access to our telephony servers.
- Students will be provided with housed computers to be used as dedicated monitoring 24/24 servers.
- We can provide to students a working space and computers in our premises during the project.
[quality] voice quality tests
- voice quality tests are difficult to do (they can be very sophisticated or quite simple; e.g. automatic comparison of source and received audio samples are complex and can be carried in collaborations with EPFL labs)
- you can relay on personal perception of several persons, you can send a voice record and publish in your web article the source audio file and the received file (provide for comparison the results of the same experiment carried out via a traditional phone line)
- you can try send faxes (source fax image, destination fax image, with a traditional line, and with a VOIP line)
- voice quality tests depend strongly on the type of codecs (which you can change on your phone). Faxes are possible only via G.711 (you can measure transmission durations)
[online_monitoring] online monitoring
CPU Load monitor of our SIP servers
Fig. 1 monitoring of the numbers of concurrent calls per server
[billing_modules] Billing
- payments to be credited to customer’s account according to telephone number available in ‘payment reason’ field of customer’s payment. A test billing server platform will be created and provided for this project
- charge customers in blocks
- create customers in blocks (with their account numbers)
[radius_asterisk] RADIUS protocol integration in Asterisk
Asterisk is an open-source private branch exchange (PBX) software.
The goal of this project is to make an Asterisk server interact with our existing VoIP infrastructure through Remote Authentication Dial In User Service (RADIUS), which manages user authentication and call accounting. The main points would be the integration of the Asterisk users in our centralized database through SIP registration, accounting of calls (internal and external), then other optional features (voicemail, vocal messages, etc.).
Main steps of the project:
- Analyze the internal RADIUS implementation of Asterisk
- Analyze the existing external developed RADIUS modules
- Report the results and study the integration with our existing infrastructure
- Install the required servers and modules (external or self-developed)
- Testing phase
Required skills:
- Independent and organized person
- Good Linux/UNIX knowledge
- Knowledge of script languages (Perl)
- Basic networking knowledge
- Notions of VoIP (SIP protocol) a plus
- Notions of SQL a plus
References:
http://www.asterisk.org/ Asterisk PBX
http://tools.ietf.org/html/rfc2865 RFC 2865 (RADIUS)
http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth External RADIUS module for Asterisk (1)
http://www-rocq.inria.fr/who/Philippe.Sultan/Asterisk/asterisk_sip_external_authentication.html External RADIUS module for Asterisk (2)
Contacts:
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