SIPSwitch Upgrade 30 Apr 2008
SIPSwitch Upgrade 30 Apr 2008
Hi All,
A major'ish upgrade has just been carried out on the sipswitch SIP Proxy software. The upgrade adds rudimentary TCP support to the sipswitch. It is now possible to register and place calls through TCP 213.200.94.182:5060. Please be aware the TCP functionality is only rudimentary at this point and incoming calls to TCP contacts are one thing that is currently not supported.
As this is a major release the unit test results for the major sipswitch components are below. To reiterate the idea behind including these results it's to help ensure that old bugs that have been fixed do not reoccur.
If anyone notices any issues they think may be related to this upgrade please post a comment to this thread.
Regards,
Aaron
A major'ish upgrade has just been carried out on the sipswitch SIP Proxy software. The upgrade adds rudimentary TCP support to the sipswitch. It is now possible to register and place calls through TCP 213.200.94.182:5060. Please be aware the TCP functionality is only rudimentary at this point and incoming calls to TCP contacts are one thing that is currently not supported.
As this is a major release the unit test results for the major sipswitch components are below. To reiterate the idea behind including these results it's to help ensure that old bugs that have been fixed do not reoccur.
If anyone notices any issues they think may be related to this upgrade please post a comment to this thread.
Regards,
Aaron
Hi TheFug,
No not yet. However once TCP support is a bit more polished TLS (which is an encrypted SIP channel) will be possible. However in this case it will only be for SIP traffic between your client and the sipswitch. Probably the more critical piece is the encryption of the RTP traffic carrying the audio between your client and your provider. The sipswitch is unlikely to ever be able to help much with that. If you have a client and can find a provider that support secure RTP you can have encrypted calls through the sipswitch right now.
Regards,
Aaron
No not yet. However once TCP support is a bit more polished TLS (which is an encrypted SIP channel) will be possible. However in this case it will only be for SIP traffic between your client and the sipswitch. Probably the more critical piece is the encryption of the RTP traffic carrying the audio between your client and your provider. The sipswitch is unlikely to ever be able to help much with that. If you have a client and can find a provider that support secure RTP you can have encrypted calls through the sipswitch right now.
Regards,
Aaron
Okay, because MySipSwitch deals only with the SIP signaling, it will only have effect on that, and since very few providers support encryption it will probaly
never happen also because of goverment rulings i guess, because just like with ADSL/internet providers, there must be logs available, to be looked into,
or at least in the near future....
I'm doing some reading about SIP signaling/networking, lately, mainly for the NAT network"features" which i want to have under control...so SIP works like it should be....(not just for the Betamax provides)
never happen also because of goverment rulings i guess, because just like with ADSL/internet providers, there must be logs available, to be looked into,
or at least in the near future....
I'm doing some reading about SIP signaling/networking, lately, mainly for the NAT network"features" which i want to have under control...so SIP works like it should be....(not just for the Betamax provides)
Thanks, The Fug.
gear: my ISP's Zyxel Modem/Router in bridge, Sitecom WL309 Router, Siemens Gigaset 301D
gear: my ISP's Zyxel Modem/Router in bridge, Sitecom WL309 Router, Siemens Gigaset 301D
Not unless you can find a SIP client multiplex the audio (RTP) and SIP onto the same port and a SIP Provider that will always send RTP from TCP port 80. I have never heard of either.hkr wrote:Can this eventually lead to VoIP made available using TCP on port 80?
There are a number of hotspots and firewalls that only allow port TCP/80.
The whole internet is a series of hacks and then hacks for hacks. In this case firewalls are hacks that were needed because hosts were not capable of securing themselves. However then firewalls cause problems for protocols like SIP (and other real-time protocols such as RTSP) so there's a hack needed to get around the first hack (the firewall). Skype is one piece of software that has done this. Skype tunnels all its traffic through port 443 if it can't get access on its default ports. This is to make it work in a firewalled corporate environment.
In the SIP World it's unlikely there wil be much effort put into trying to circumvent firewalls. Most of the SIP standards and SIP equipment is manufactured are contributed to or manufactured by corporates who are the same people running the firewalls. That's not to say it's right or wrong that's just the situation.
If tunnelling SIP calls through a corporate firewall is something that is a big deal for you then it is possible to do a similar thing to what Skype has done with a custom VPN. OpenVPN is one product I was able to get SIP calls working over port 443 with but the quality was very poor and the configuration effort high so I wouldn't recommend going down that path. Installing Skype would be a lot easier if the ultimate goal is cheaper calls.
Actually I've never investigated the Google IM client, maybe it can tunnel thorugh port 443. There are gTalk-to-SIP gateways so if it can do the tunnel that would be an avenue from a corporate network out to the SIP one.
Regards,
Aaron
I guess, even if you should have sip or the media encrypted it's also an extra, where things could go wrong, i guess one should be happy, having voip just working, in a save envourment, most of the time, this is also a form of security, one is not far from the other and wishing for IPv6
Thanks, The Fug.
gear: my ISP's Zyxel Modem/Router in bridge, Sitecom WL309 Router, Siemens Gigaset 301D
gear: my ISP's Zyxel Modem/Router in bridge, Sitecom WL309 Router, Siemens Gigaset 301D