--- channels/chan_sip.c (Asterisk 16.8-cert2)
+++ channels/chan_sip.c (working copy)
@@ -34298,6 +34298,39 @@
 		return FALSE;
 	}
 
+	/* Nokia Symbian/S60 requires little-endian ROC; default of RFC is network
+	 * order = big endian. Nokia Series 40 and Nokia Asha Software Platform are
+	 * not affected and use network order. As User-Agent, these use NokiaModel,
+	 * for example Nokia6300i and Nokia503. Nokia MeeGo/Maemo do not offer sRTP.
+	 * Symbian/S60 uses *<space>Type, Type is RA-x or RM-xxx, for example:
+	 * a) Nokia RA-6 300.34.84
+	 * b) E66-1 RM-343 510.21.009
+	 * c) S60 RM-670 113.010.1506
+	 *
+	 * http://www.nokiaport.de/?pid=models
+	 * http://web.archive.org/web/20080416025322/http://www.forum.nokia.com/main/resources/technologies/voice_over_IP/voip_support_in_nokia_devices.html
+	 * http://web.archive.org/web/20090528115458/http://www.forum.nokia.com/Technology_Topics/Mobile_Technologies/VoIP/Nokia_VoIP_Framework/VoIP_support_in_Nokia_devices.xhtml
+	 * http://web.archive.org/web/20120309140131/http://www.developer.nokia.com/Community/Wiki/VoIP_support_in_Nokia_devices
+	 * Nokia E63, Nokia E66, and Nokia E71 (only those) send no User-Agent on
+	 * REGISTER and no while answering to an INVITE. Therefore, these phones
+	 * cannot be catched/filtered by the code below. Calls initiated by these
+	 * phones do work because the User-Agent is sent. If you have users using
+	 * such phones to receive calls, consider to change the rtp->seqno from
+	 * random to 0 (in res/res_rtp_asterisk.c:ast_rtp_new); then calls last at
+	 * least 21 minutes.
+	 */
+	if (strstr(p->useragent, " RA-")
+	 || strstr(p->useragent, " RM-")
+	 /* required for INVITE from Asterisk because INVITE session does not exchange User-Agent: */
+	 || (p->relatedpeer && strstr(p->relatedpeer->useragent, " RA-"))
+	 || (p->relatedpeer && strstr(p->relatedpeer->useragent, " RM-"))) {
+		struct ast_sdp_srtp *tmp = *srtp;
+		do {
+			ast_set_flag(tmp, AST_SRTP_CRYPTO_ROC_LE);
+			tmp = AST_LIST_NEXT(tmp, sdp_srtp_list);
+		} while (tmp);
+	}
+
 	if (ast_sdp_crypto_process(rtp, *srtp, a) < 0) {
 		return FALSE;
 	}
--- include/asterisk/res_srtp.h	(Asterisk 16.8-cert2)
+++ include/asterisk/res_srtp.h	(working copy)
@@ -55,4 +55,6 @@
 /* Crypto suites */
 enum ast_srtp_suite {
+	AST_AES_CM_128_HMAC_SHA1_32_ROC_LE = -2,
+	AST_AES_CM_128_HMAC_SHA1_80_ROC_LE = -1,
 	/* https://www.iana.org/assignments/srtp-protection/srtp-protection.xhtml */
 	AST_AES_CM_128_HMAC_SHA1_80 = 1,
--- include/asterisk/sdp_srtp.h (Asterisk 16.8-cert2)
+++ include/asterisk/sdp_srtp.h (working copy)
@@ -42,2 +42,3 @@
 
+#define AST_SRTP_CRYPTO_ROC_LE   (1 << 9)
 /* SRTP flags */
--- res/res_srtp.c (Asterisk 16.8-cert2)
+++ res/res_srtp.c (working copy)
@@ -319,6 +319,13 @@
 
 static int ast_srtp_policy_set_suite(struct ast_srtp_policy *policy, enum ast_srtp_suite suite)
 {
+	if (suite == AST_AES_CM_128_HMAC_SHA1_32_ROC_LE) {
+		policy->sp.allow_repeat_tx = (policy->sp.allow_repeat_tx | 0x02);
+		suite = AST_AES_CM_128_HMAC_SHA1_32;
+	} else if (suite == AST_AES_CM_128_HMAC_SHA1_80_ROC_LE) {
+		policy->sp.allow_repeat_tx = (policy->sp.allow_repeat_tx | 0x02);
+		suite = AST_AES_CM_128_HMAC_SHA1_80;
+	}
 	return policy_set_suite(&policy->sp.rtp, suite) | policy_set_suite(&policy->sp.rtcp, suite);
 }
 
@@ -867,2 +874,5 @@
 		suite_val = AST_AES_CM_128_HMAC_SHA1_80;
+		if (ast_test_flag(srtp, AST_SRTP_CRYPTO_ROC_LE)) {
+			suite_val = AST_AES_CM_128_HMAC_SHA1_80_ROC_LE;
+		}
 		ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_80);
@@ -871,2 +881,5 @@
 		suite_val = AST_AES_CM_128_HMAC_SHA1_32;
+		if (ast_test_flag(srtp, AST_SRTP_CRYPTO_ROC_LE)) {
+			suite_val = AST_AES_CM_128_HMAC_SHA1_32_ROC_LE;
+		}
 		ast_set_flag(srtp, AST_SRTP_CRYPTO_TAG_32);