[cisco-voip] sRTP and RTP in SIP Invite

Brian Meade bmeade90 at vt.edu
Fri May 30 12:20:00 EDT 2014


Can you send a CallManager SDI/SDL trace for one of these calls?


On Fri, May 30, 2014 at 12:14 PM, Mark Holloway <mh at markholloway.com> wrote:

> Yep, it’s TLS.  Certificates are loaded.
>
>
> On May 30, 2014, at 11:48 AM, Brian Meade <bmeade90 at vt.edu> wrote:
>
> Mark,
>
> Is the device actually using TLS for the signaling?  I don't think CUCM
> will let you use SRTP if the signaling channel isn't encrypted.
>
> Brian
>
>
> On Fri, May 30, 2014 at 11:41 AM, Mark Holloway <mh at markholloway.com>
> wrote:
>
>> I’ve got a non-Cisco SIP device sending SIP Invites to CUCM (SIP Trunk).
>> The SDP from my device includes RTP and sRTP in the SIP Invite. Reading
>> Cisco docs it looks like the way Cisco expects sRTP to work is the SIP
>> Invite should only include sRTP assuming if the call should be encrypted.
>>  If both RTP and sRTP are in the SDP, CUCM will always choose the first one
>> in the list rather than the preferred type (sRTP in this example).  In my
>> case RTP is being listed first then sRTP, therefore CUCM will never choose
>> sRTP even though that is what I prefer.
>>
>> Has anyone encountered this before and is there a way around it?
>>
>> Thanks,
>> Mark
>>
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>
>
>
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