Maybe I misunderstand your question but are you sure you problem is not the acc->cfg.srtp_secure_signaling setting?
If that is set to require a secure signalling transport when SRTP is used, changing the value of pjsua_acc_config.use_srtp is probably not going to help if you use a non secure address.
IIRC pjsua_acc_config.use_srtp basically checks if the sdp info contains RTP/AVP or RTP/SAVP and PJMEDIA_SRTP_OPTIONAL would allow both to pass but that is not related to the sip vs sips in the signalling part.
Johan
On 07/04/2017, 18:46, "pjsip on behalf of David Talmage" <pjsip-bounces@xxxxxxxxxxxxxxx on behalf of sip.phone.fan@xxxxxxxxx> wrote:
>I think I found a bug in the way that PJSIP handles the SRTP settings.
>Would someone please confirm this?
>
>When pjsua_acc_config.use_srtp is not PJMEDIA_SRTP_DISABLED, PJSIP
>requires SRTP and all calls must be addressed to a sips: URI. PJSIP
>rejects calls to sip: URIs with the status code
>PJSIP_ESESSIONINSECURE.
>
>The behavior I expect is for PJSIP to fall back to an insecure call
>when the destination URI is sip: and the value of
>pjsua_acc_config.use_srtp is PJMEDIA_SRTP_OPTIONAL.
>
>The mistake is in pjsua_media.c:call_media_init_cb(). Here is the code:
>
> /* Check if SRTP requires secure signaling */
> if (acc->cfg.use_srtp != PJMEDIA_SRTP_DISABLED) {
> if (security_level < acc->cfg.srtp_secure_signaling) {
> err_code = PJSIP_SC_NOT_ACCEPTABLE;
> status = PJSIP_ESESSIONINSECURE;
> goto on_return;
> }
> }
>
>I don't have a working solution yet. It looks easy but perhaps there
>will be unintended consequences.
>
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