Re: [asterisk-dev] s?rtp via SIP/TLS/TCP

2015-04-22 Thread Olle E. Johansson
On 21 Apr 2015, at 17:55, James Cloos wrote: >> "OEJ" == Olle E Johansson writes: > > OEJ> It's a bug in chan_sip that I fixed a while ago in one of my branches. > OEJ> SNOM sends an SDES key but RTP/AVP in the offer and chan_sip > OEJ> chokes. It's a one or two line fix - or turn off SRTP

Re: [asterisk-dev] s?rtp via SIP/TLS/TCP

2015-04-21 Thread James Cloos
> "OEJ" == Olle E Johansson writes: OEJ> It's a bug in chan_sip that I fixed a while ago in one of my branches. OEJ> SNOM sends an SDES key but RTP/AVP in the offer and chan_sip OEJ> chokes. It's a one or two line fix - or turn off SRTP in the SNOM. I presume this one?: sdes-rtp-avp.diff: I

Re: [asterisk-dev] s?rtp via SIP/TLS/TCP

2015-04-21 Thread James Cloos
> "MJ" == Matthew Jordan writes: MJ> Guessing at what is occurring here: your phone is probably offering MJ> optional/optimistic encryption. While optimistic encryption is MJ> supported by chan_pjsip, currently, an offer with RTP/AVPF with crypto MJ> attributes is currently rejected by chan_s

Re: [asterisk-dev] s?rtp via SIP/TLS/TCP

2015-04-20 Thread Olle E. Johansson
On 20 Apr 2015, at 17:41, James Cloos wrote: > I'm not sure whether this is a bug, so I'm starting here. > > My remote asterisk (debian's compile of 13, currently 13.1.0) and my > snom had been unable to rtp for some time. I still use chan_sip. > > It took a few hours of testing, but I determ

Re: [asterisk-dev] s?rtp via SIP/TLS/TCP

2015-04-20 Thread Matthew Jordan
On Mon, Apr 20, 2015 at 10:41 AM, James Cloos wrote: > I'm not sure whether this is a bug, so I'm starting here. > > My remote asterisk (debian's compile of 13, currently 13.1.0) and my > snom had been unable to rtp for some time. I still use chan_sip. > > It took a few hours of testing, but I de

[asterisk-dev] s?rtp via SIP/TLS/TCP

2015-04-20 Thread James Cloos
I'm not sure whether this is a bug, so I'm starting here. My remote asterisk (debian's compile of 13, currently 13.1.0) and my snom had been unable to rtp for some time. I still use chan_sip. It took a few hours of testing, but I determined that when the phone registers and/or invites over tls,