Hi,
I updated to the latest from git but it didn't improve the situation with null IP addresses from the non-webRTC client. In fact, rtpengine sent some ICE candidate lines with null IP's. We decided we don't need to worry about clients that use the null IP mechanism for on-hold, so I sent some simulated traffic with sipp that has real IP addresses and uses a=sendonly/inactive. These results were better.
With the latest rtpengine version (as of 21/7/14), the initial invite towards Chrome uses a=setup:actpass for the DTLS handshake, as required. In the reINVITE, it uses a=setup:passive. Chrome rejects this because it isn't actpass. I can see the code that works out the setup direction string in sdp.c:1600 based on the active/passive flags. Should the value always be "actpass" if we are making an offer? (RFC5763 ยง5)
Regards, Hugh
-----Original Message----- From: sr-dev-bounces@lists.sip-router.org [mailto:sr-dev-bounces@lists.sip-router.org] On Behalf Of Richard Fuchs Sent: 17 July 2014 18:06 To: sr-dev@lists.sip-router.org Subject: Re: [sr-dev] RTPEngine in reINVITE/on-hold use case
On 07/17/14 12:48, Waite, Hugh wrote:
Hi,
I am using rtpengine to convert SRTP to RTP for audio and video and I am also doing this for reINVITEs.
RTPEngine appears to have disabled the two media streams, rather than putting them on hold. Is this a deliberate choice or the best choice given the multiple possibilities of client implementations? And can this behaviour be improved?
If the call is taken off-hold later, I would prefer not to have to do the full ICE restart, including TURN, DTLS handshakes etc. etc. if it can be avoided.
This is one of those cases where no matter what you do, it's always gonna be wrong for somebody. :)
There was a recent commit [1] which slightly changed the behaviour when putting a call on hold in this manner, I suspect your build is older than this. I'm not sure if it would fix your particular issue though.
Rtpengine could translate a null c= address into the respective attributes (and leave a valid c= address), but then you run into the problem of breaking compatibility with clients which don't support or misunderstand this type of signalling.
So yes, there's definitely room for improvement (there always is), it's just a matter of figuring out what the Right Thing[tm] to do is. :)
cheers
[1] https://github.com/sipwise/rtpengine/commit/a7784f5ca3437d1c7b58308022ef0e06...
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