测试 rtpengine 的 sip-source 参数

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已于 2024-04-17 14:27:42 修改

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分类专栏： rtpengine 文章标签：通讯

于 2024-04-16 16:56:32 首次发布

本文链接：https://blog.csdn.net/qq_56167818/article/details/137831202

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修改 /etc/rtpengine/rtpengine.conf 文件，增加一行配置：

sip-source = true # 一般不需要这样配置，本文仅为说明问题

offfer 部分的日志如下：

[1713246486.390578] DEBUG: [1-5025@192.168.43.126]: [control] Dump for 'offer' from 127.0.0.1:34943: { "supports": [ "load limit" ], "sdp": "v=0
o=XRMG 1704250570 3 IN IP4 10.143.1.14
s=XRMG V2.0
c=IN IP4 10.143.1.14
t=0 0
m=audio 30608 RTP/AVP 8 101
c=IN IP4 10.143.1.14
b=AS:82
b=RS:600
b=RR:2000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
", "replace": [ "origin", "session-connection" ], "call-id": "1-5025@192.168.43.126", "via-branch": "z9hG4bK-5025-1-00", "received-from": [ "IP4", "192.168.43.126" ], "from-tag": "5025SIPpTag001", "command": "offer" }

这里 sip-source 的地址（来自 received-from）为 192.168.43.126，端口来自 sdp 里面的 30608

接下来的日志为：

[core] set FILLED flag for stream, local 192.168.43.126:36912 remote 192.168.43.126:30608

远端地址不是 sdp 的 c 地址，而是 sip-source

源码里面 trust_address 如果为 true 代表采用 sdp 的 c 地址，跟 sip-source 的意思正好是反过来的

// call_interfaces.c

// ...

static void call_ng_flags_flags(struct sdp_ng_flags *out, str *s, void *dummy) {
	str_hyphenate(s);

	switch (__csh_lookup(s)) {
		case CSH_LOOKUP("trust-address"):
			out->trust_address = 1;   // sdp c address
			break;
		case CSH_LOOKUP("SIP-source-address"):
		case CSH_LOOKUP("sip-source-address"):
			out->trust_address = 0;  // sip-source
			break;

// sdp.c
// ...
static int fill_endpoint(struct endpoint *ep, const struct sdp_media *media, struct sdp_ng_flags *flags,
		struct network_address *address, long int port)
{
	struct sdp_session *session = media->session;

	if (!flags->trust_address) {
		if (is_addr_unspecified(&flags->parsed_received_from)) {
			if (__parse_address(&flags->parsed_received_from, NULL, &flags->received_from_family,
						&flags->received_from_address))
				return -1;
		}
		ep->address = flags->parsed_received_from; // sip-source
	}

目前版本的 rtpengine 默认的配置就是 trust-address，不再是 sip-source

熟悉 kamailio 的都知道，havfo 的脚本牛的很，现在可以解读一下了：

request_route {
   # per request initial checks
   route(REQINIT);

   if (nat_uac_test(64)) {
       # Do NAT traversal stuff for requests from a WebSocket
       # connection - even if it is not behind a NAT!
       # This won't be needed in the future if Kamailio and the
       # WebSocket client support Outbound and Path.
       force_rport();
       if (is_method("REGISTER")) {
           fix_nated_register();
       } else if (!add_contact_alias()) {
           xlog("L_ERR", "Error aliasing contact <$ct>\n");
           sl_send_reply("400", "Bad Request");
           exit;
       }
   }

# nat_uac_test(64) 就是测试信令是不是 WebSocket

# 如果是 WebSocket，那么需要 force_rport, 因为 Via 地址根本不可达

route[NATMANAGE] {

# ...

if (!nat_uac_test("8")) {
$xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " trust-address";
}

# nat_uac_test("8") 测试sdp里面的地址是不是 rfc1918

# 如果不是 rfc1918，那么 trust-address，当时写这个脚本的时候，rtpengine 的版本应该比较旧

onreply_route {
   if ((($Rp == MY_WS_PORT || $Rp == MY_WSS_PORT)
       && !(proto == WS || proto == WSS))) {
       xlog("L_WARN", "SIP response received on $Rp\n");
       drop;
   }

   if (nat_uac_test(64)) {
       # Do NAT traversal stuff for replies to a WebSocket connection
       # - even if it is not behind a NAT!
       # This won't be needed in the future if Kamailio and the
       # WebSocket client support Outbound and Path.
       add_contact_alias();
   }
}

# reply 路由块的处理

# 先检查接收端口

# 再做 nat_uac_test(64) 检查，也就是终端是不是 WebSocket

# 如果是，那么需要 rewrite contact, 因为 WebSocket 的 contact 地址根本不可达