sipcall: DTMF over RTP is supported

pjsip supports RFC 4733 (RTP Payload for DTMF Digits)
So, DTMF over RTP is supported.
Source: https://trac.pjsip.org/repos/wiki/PJSIP-Datasheet

Change-Id: I982cf20a48f5919d5ee62aac13a7c9daaabd7296
GitLab: #9

src/sip/sipcall.cpp

@@ -84,14 +84,6 @@ const char* const SIPCall::LINK_TYPE = SIPAccount::ACCOUNT_TYPE;
 static void
 dtmfSend(SIPCall &call, char code, const std::string &dtmf)
 {
-    if (dtmf == SIPAccount::OVERRTP_STR) {
-        RING_WARN("[call:%s] DTMF over RTP not supported yet", call.getCallId().c_str());
-        return;
-    } else if (dtmf != SIPAccount::SIPINFO_STR) {
-        RING_WARN("[call:%s] Unknown DTMF type %s, defaulting to %s instead",
-                  call.getCallId().c_str(), dtmf.c_str(), SIPAccount::SIPINFO_STR);
-    } // else : dtmf == SIPINFO
-
     int duration = Manager::instance().voipPreferences.getPulseLength();
     char dtmf_body[1000];
 

[Timur Guzairov]

But how can we access pjsua functionality from jami-daemon? Especially

pjsua_call_dial_dtmf

function? So, even if pjsip supports RTP Payload for DTMF Digits, Jami itself doesn't. It supports only DTMF using INFO

[Sébastien Blin]

By changing the code to use it