ah ok, looking closely at RTPEngine logs, i also see this message,

--
rtpengine[16455]: [j4ujoo87lsnjvapf4cmd port 41035] SRTCP output wanted, but no crypto suite was negotiated
--

So you are right about crypto mismatch, however it is strange why SRTCP was not negotiated with chrome ...

Thank you.



On Wed, Feb 4, 2015 at 10:17 PM, Muhammad Shahzad <shaheryarkh@gmail.com> wrote:
For the moment the message appears when a webrtc caller (using chrome v40.0.2214.94 64-bit) dials voicemail service which is running on a legacy sip server. The RTPEngine converts media from SAVPF to AVP for this audio only call.

There may be other scenarios, since this is not the first time i see this message. I will update here when i encounter those scenarios again.

Thank you.



On Wed, Feb 4, 2015 at 9:59 PM, Richard Fuchs <rfuchs@sipwise.com> wrote:
On 02/04/15 15:42, Muhammad Shahzad wrote:
> Hi,
>
> I have latest stable release of RTPEngine deployed in a virtual machine
> (KVM) along with Kamailio v4.2. All is working fine except i see this
> message in RTPEngine logs,
>
> --
> rtpengine[16455]: [82qrjq0hdtt45afbqo98 port 40960] Kernelizing media
> stream
> rtpengine[16455]: [82qrjq0hdtt45afbqo98 port 40960] No support for
> kernel packet forwarding available
> --

This message pops up if there's an unsupported combination of crypto
suites or transport protocols being used. I'm not aware of any such
combination in the current code base. Can you provide more details about
what kind of calls you're handling there?

cheers

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