wg-voip - Re: More on SIP hard phones
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- From: Andrew Rutherford <>
- To: Ben Teitelbaum <>, I2 VoIP working group <>
- Cc: <>
- Subject: Re: More on SIP hard phones
- Date: Fri, 10 Oct 2003 09:51:24 +0930
At 1:26 PM -0400 8/10/03, Ben Teitelbaum wrote:
Are we talking about a matrix of {phones} X {proxies} or list of SIP
gear or all sorts, or what? Would the content be based on anecdotal
reports, official vendor claims, or formal interop testing?
Unfortunately it's not always quite that easy. Often the problems are end-device to end-device, but the proxy server can try and mitigate the issues.
Classic case: When using G.729, Cisco IOS devices advertise the codec as:
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
But if Cisco's 79[46]0 phones receive this in an INVITE, and there is a lower codec in the list, they ignore the G.729 codec and fall through to the lower codecs unless the proxy server removes either the "fmtp" line or the lower codecs, in which case the phone will quite happily accept G.729.
(Yes, preferred_codec is set to "none", if anyone else is familiar with that fix for standards breaches in those phones.)
Quite often we see proxy servers trying to fix up things so implementations can talk to each other. Other common ones include changing loose source routing (RFC 3261) to strict source routing (RFC 2543) for devices that will reject inbound calls if the newer form of routing is used, such as the ubiquitous Windows XP Instant Messenger.
So the matrix could actually be {phones} X {proxies} X {phones}, which isn't fun to represent. I'd suggest instead doing {phones} X {phones}, specifying "yes", "with fixups A, B, and C", or "no", and then listing the proxies with what fixups they do.
There's also another class of issues when the chain gets four devices long instead of the typical LAN environment of three - that is, phone talks to local proxy server, which talks to target proxy server, which talks to target phone - the typical long-distance environment.
We see all kinds of fun things when one proxy server uses strict routing and the other uses loose routing, and one of the devices in the chain doesn't handle the mix right and tries to route to the second device (which it can't reach because of security policy) instead of the first. In one case, a lab test was done and they didn't pick it - the INVITE and 200 worked, but the ACK didn't get through after the media stream was set up - so the phone timed out the call on not getting the ACK after 30 seconds - but none of their test calls went to 30 seconds so they never noticed it. :-(
--
Andrew Rutherford
sip:
244 Pirie Street
Iagu Networks tel:+61-8-8425-2255 Adelaide SA 5000
http://www.iagu.net/
mailto:
Australia
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- More on SIP hard phones, Chris Peabody, 10/03/2003
- Re: More on SIP hard phones, Steve Blair, 10/03/2003
- Re: More on SIP hard phones, Jiri Kuthan, 10/06/2003
- <Possible follow-up(s)>
- Re: More on SIP hard phones, peabodyc, 10/04/2003
- Re: More on SIP hard phones, Ben Teitelbaum, 10/08/2003
- Re: More on SIP hard phones, Andrew Rutherford, 10/09/2003
- Re: More on SIP hard phones, Ben Teitelbaum, 10/08/2003
- RE: More on SIP hard phones, Barry Wray, 10/09/2003
- RE: More on SIP hard phones, janar, 10/14/2003
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