Tag Archives: sip

BT unleashes SIP licensing troll army

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VoIP-to-PSTN termination providers and SIP vendors will be watching their inboxes for a lawyer’s letter from BT, which has kicked off a taxing licensing program levying a fee on the industry, based on a list of 99 patents.

via BT unleashes SIP licensing troll army • The Register.

A useful comment from slashdot.

The IETF MMUSIC (Multiparty Multimedia Session Control) Working Group started working on Session Protocols [ietf.org] in 1993.

Initial Internet drafts for a Session Invitation Protocol and a Simple Conference Invitation Protocol were prepared in 1996, and merged to a single first draft of SIP by December 1996 (slide 10 [columbia.edu]), with further drafts (2-12) leading up to the publication of RFC 2543 in March of 1999 (slides 11-13, ibid.).

I don’t see anything that says BT had a hand in anything to do with SIP up to 1996. More than half the patents BT claims (Exhibit C [btplc.com]) were filed after RFC 2543 was published.

I hope this information is a useful starting point for some SIP vendor.

VoIPmonitor

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VoIPmonitor is open source network packet sniffer with commercial frontend for SIP RTP and RTCP VoIP protocols running on linux. VoIPmonitor is designed to analyze quality of VoIP call based on network parameters – delay variation and packet loss according to ITU-T G.107 E-model which predicts quality on MOS scale. Calls with all relevant statistics are saved to MySQL or ODBC database. Optionally each call can be saved to pcap file with either only SIP protocol or SIP/RTP/RTCP/T.38/udptl protocols. VoIPmonitor can also decode speech and play it over the commercial WEB GUI or save it to disk as WAV. Supported codecs are G.711 alaw/ulaw and commercial plugins supports G.722 G.729a G.723 iLBC Speex GSM Silk iSAC. VoIPmonitor is also able to convert T.38 FAX to PDF.

via VoIPmonitor – VoIP monitoring software – quality analyzer – WAV recorder.

Kamailio SIP Server

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Kamailio™ (former OpenSER) is an Open Source SIP Server released under GPL, able to handle thousands of call setups per second. Among features: asynchronous TCP, UDP and SCTP, secure communication via TLS for VoIP (voice, video); IPv4 and IPv6; SIMPLE instant messaging and presence with embedded XCAP server and MSRP relay; ENUM; DID and least cost routing; load balancing; routing fail-over; accounting, authentication and authorization; support for many backend systems such as MySQL, Postgres, Oracle, Radius, LDAP, Redis, Cassandra; XMLRPC control interface, SNMP monitoring. It can be used to build large VoIP servicing platforms or to scale up SIP-to-PSTN gateways, PBX systems or media servers like Asterisk™, FreeSWITCH™ or SEMS. Kamailio and the SIP Express Router (SER) teamed up for the integration of the two applications and new development.

via Kamailio SIP Server.

Welcome to SIPp

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SIPp is a free Open Source test tool / traffic generator for the SIP protocol. It includes a few basic SipStone user agent scenarios (UAC and UAS) and establishes and releases multiple calls with the INVITE and BYE methods. It can also reads custom XML scenario files describing from very simple to complex call flows. It features the dynamic display of statistics about running tests (call rate, round trip delay, and message statistics), periodic CSV statistics dumps, TCP and UDP over multiple sockets or multiplexed with retransmission management and dynamically adjustable call rates.

via Welcome to SIPp.

SIPp can be used to test many real SIP equipements like SIP proxies, B2BUAs, SIP media servers, SIP/x gateways, SIP PBX, … It is also very useful to emulate thousands of user agents calling your SIP system.

PBXMate-FreeSWITCH-integration

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The PBXMate software product from SoliCall is designed to improve voice quality by canceling echo, canceling noise and monitoring quality indicators. This article describes, in details, one option to integrating the PBXMate with FreeSWITCH in which both products are installed on the same Linux machine and a DNS is being used.

via PBXMate-FreeSWITCH-integration – FreeSWITCH Wiki.

SIP Trunking: A Reality Check

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As the conversation developed, we learned there are companies that are not planning, testing and monitoring installations. They are simply putting systems in, getting SIP trunks connected, testing for dial tone and moving onto the next client. Customers are experiencing poor quality, dropped calls and SIP trunks simultaneously dropping then reconnecting. So many problems occurred, the customers simply said, “Enough is enough! Let’s rip everything out and go back to what works!”

via SIP Trunking: A Reality Check.