tested with Asterisk 11.17.1
--- include/asterisk/res_srtp.h	(revision 371121)
+++ include/asterisk/res_srtp.h	(working copy)
@@ -52,6 +52,8 @@
 
 /* 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,
 	AST_AES_CM_128_HMAC_SHA1_80 = 1,
 	AST_AES_CM_128_HMAC_SHA1_32 = 2,
 	AST_F8_128_HMAC_SHA1_80     = 3
--- channels/chan_sip.c	(revision 434288)
+++ channels/chan_sip.c	(working copy)
@@ -33402,6 +33402,7 @@
 
 static int process_crypto(struct sip_pvt *p, struct ast_rtp_instance *rtp, struct sip_srtp **srtp, const char *a)
 {
+	unsigned int roll_over_counter_little_endian = 0;
 	struct ast_rtp_engine_dtls *dtls;
 
 	/* If no RTP instance exists for this media stream don't bother processing the crypto line */
@@ -33429,7 +33430,39 @@
 		return FALSE;
 	}
 
-	if (sdp_crypto_process((*srtp)->crypto, a, rtp, *srtp) < 0) {
+	/* 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-")) {
+		roll_over_counter_little_endian = 1;
+	} else if (strstr(p->useragent, " RM-")) {
+		roll_over_counter_little_endian = 1;
+	/* required for INVITE from Asterisk because session does not exchange User-Agent */
+	} else if (p->relatedpeer && strstr(p->relatedpeer->useragent, " RA-")) {
+		roll_over_counter_little_endian = 1;
+	} else if (p->relatedpeer && strstr(p->relatedpeer->useragent, " RM-")) {
+		roll_over_counter_little_endian = 1;
+	}
+
+	if (sdp_crypto_process((*srtp)->crypto, a, rtp, *srtp, roll_over_counter_little_endian) < 0) {
 		return FALSE;
 	}
 
--- channels/sip/include/sdp_crypto.h	(revision 371121)
+++ channels/sip/include/sdp_crypto.h	(working copy)
@@ -57,7 +57,7 @@
  * \retval 0 success
  * \retval nonzero failure
  */
-int sdp_crypto_process(struct sdp_crypto *p, const char *attr, struct ast_rtp_instance *rtp, struct sip_srtp *srtp);
+int sdp_crypto_process(struct sdp_crypto *p, const char *attr, struct ast_rtp_instance *rtp, struct sip_srtp *srtp, unsigned int roc_le);
 
 
 /*! \brief Generate an SRTP a=crypto offer
--- channels/sip/sdp_crypto.c	(revision 432239)
+++ channels/sip/sdp_crypto.c	(working copy)
@@ -185,7 +185,7 @@
 	return res;
 }
 
-int sdp_crypto_process(struct sdp_crypto *p, const char *attr, struct ast_rtp_instance *rtp, struct sip_srtp *srtp)
+int sdp_crypto_process(struct sdp_crypto *p, const char *attr, struct ast_rtp_instance *rtp, struct sip_srtp *srtp, unsigned int roc_le)
 {
 	char *str = NULL;
 	char *tag = NULL;
@@ -232,10 +232,16 @@
 
 	if (!strcmp(suite, "AES_CM_128_HMAC_SHA1_80")) {
 		suite_val = AST_AES_CM_128_HMAC_SHA1_80;
+		if (roc_le) {
+			suite_val = AST_AES_CM_128_HMAC_SHA1_80_ROC_LE;
+		}
 		ast_set_flag(srtp, SRTP_CRYPTO_TAG_80);
 		taglen = 80;
 	} else if (!strcmp(suite, "AES_CM_128_HMAC_SHA1_32")) {
 		suite_val = AST_AES_CM_128_HMAC_SHA1_32;
+		if (roc_le) {
+			suite_val = AST_AES_CM_128_HMAC_SHA1_32_ROC_LE;
+		}
 		ast_set_flag(srtp, SRTP_CRYPTO_TAG_32);
 		taglen = 32;
 	} else {
--- res/res_srtp.c	(revision 426141)
+++ res/res_srtp.c	(working copy)
@@ -278,6 +278,13 @@
 
 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);
 }