--- /usr/local/src/orig/asterisk-1.4.33.1/channels/chan_sip.c   2010-06-01 22:57:49.000000000 +0800
+++ /usr/local/src/asterisk-1.4.33.1/channels/chan_sip.c        2010-07-28 16:58:21.000000000 +0800
@@ -6763,14 +6763,23 @@

        if (debug)
                ast_verbose("Adding codec 0x%x (%s) to SDP\n", codec, ast_getformatname(codec));
-       if ((rtp_code = ast_rtp_lookup_code(p->rtp, 1, codec)) == -1)
+
+       struct ast_rtp* rtp = NULL;
+
+       if (codec & AST_FORMAT_AUDIO_MASK)
+               rtp = p->rtp;
+       if (codec & AST_FORMAT_VIDEO_MASK)
+               rtp = p->vrtp;
+
+       if(!rtp)
                return;

-       if (p->rtp) {
-               struct ast_codec_pref *pref = ast_rtp_codec_getpref(p->rtp);
-               fmt = ast_codec_pref_getsize(pref, codec);
-       } else /* I dont see how you couldn't have p->rtp, but good to check for and error out if not there like earlier code */
+       if ((rtp_code = ast_rtp_lookup_code(rtp, 1, codec)) == -1)
                return;
+
+       struct ast_codec_pref *pref = ast_rtp_codec_getpref(rtp);
+       fmt = ast_codec_pref_getsize(pref, codec);
+
        ast_build_string(m_buf, m_size, " %d", rtp_code);
        ast_build_string(a_buf, a_size, "a=rtpmap:%d %s/%d\r\n", rtp_code,
                         ast_rtp_lookup_mime_subtype(1, codec,