2004 March

March 2004

\"\"This morning at Von 2004 Jeff Pulver delivered a keynote which can be summarized in a few bullets:

  • Be innovative!
  • Be passionate!
  • IP communication is much more than a PSTN replacement service
  • Help the community educate the regulators

I think this are important messages to the VoIP community. Real time IP communication needs to be much more than simple phone calls over packet networks. The real profit is not in low-priced services, it\’s in value-added services that customers can benefit from in a way that will help them raise their profit.

is an interesting mixture of politics, technology, business strategies and visions. Lot to learn, lot of people to meet. A huge brainstorm in Santa Clara!

Tonight, the build process for Asterisk was changed so that the old IAX channel that only supported IAX version 1 was taken out of the build process. If you need it for backwards compatibility, you need to change the makefile in the \”channels\” directory. This affects
bot CVS versions and will be part of the next release.

Maxim Sobolev, PortaOne, has released a new version of the PortaOne RTP proxy and the SIP Express Router NAT-helper module that enables IPv4 to IPv6 transition for SIP networks. The combination enables the SIP Express Router to act as an application level gateway for SIP networks with users both on IPv4 and IPv6. The RTPproxy will now carry the media stream between the different address spaces, acting in the same way as it does when bridging calls between private networks with NAT.
This is a major improvement in enabling VoIP calls across the two IP worlds.

Yesterday Asterisk got an update to the queue system. The queue system is now able to deliver messages saying \”The estimated waiting time is 3 minutes\”, \”You are first in line\” and \”There are five calls waiting\”, giving your company a more professional queue system for customers waiting to reach your staff. This is one of the additions in preparation of Asterisk version 0.9.0.

Finally, pictures of the IAX ATA from Digium:

The IAXy (pronounced \”eek-see\”) is a single-port Ethernet to Telephone adapter designed for use with Asterisk, Digium\’s open-source telephony software.
Unlike all other ATA devices to date, IAXy, as its name implies, uses IAX (the Inter Asterisk eXchange protocol), rather than SIP, for its communications on the wire.

SIP digest authentication is widely misunderstood. It consists of four basic parts:

  • The user you authenticate as
  • The domain the user belongs to
  • The realm you authenticate to
  • The secret you use to authenticate

The realm is a unique identifier for the protection space, it could be Mycompany PSTN gateway.
The username and domain is the authentication you are given for this protection space. In most cases, this is not the same as your SIP address, the sip:username@sipdomain address you use. One SIP address could have multiple authentications into different realms. The SIP user could have one authentication for the company domain SIP registrar service. Another one for a commercial SIP pstn service and a third for an outbound SIP proxy in the hotel.

In most cases, this is not possible in today\’s phones. This needs to be fixed.
Take the popular Grandstream phone. It\’s even impossible to have a SIP address like sip:username@sipdomain and register with a proxy with another name, like proxy.proxydomain.tld. The Grandstream phones always register with the proxy name taken
from the domain part of the SIP address. Wish they would implement lookup of DNS srv/naptr records, wish they would implement proper SIP digest authentication support. Xten\’s Xlite software is almost there. You can configure it authentication realm and user separately from SIP username and domain, but as I understand, it doesn\’t support multiple realms for one user.

Customers and service providers need to put pressure on the vendors to do things right,
to follow the standards. And we need to hurry, before we have a massive installed base that
can\’t easily be fixed.