librtmp推流问题

Question

使用librtmp库进行推流，往虎牙直播推流正常，往斗鱼直播推流时提示无效的推流码，推流失败，不知道什么原因，librtmp打印如下：

[RTMP]Connect
DEBUG: Parsing...
DEBUG: Parsed protocol: 0
DEBUG: Parsed host    : 172.16.172.109
DEBUG: Parsed app     : live
==>palypath0:dystream?time=1748069363
==>palypath:dystream?time=1748069363
DEBUG: RTMP_Connect1, ... connected, handshaking
DEBUG: HandShake: Type Answer   : 03
DEBUG: HandShake: Server Uptime : 1748072128
DEBUG: HandShake: FMS Version   : 0.0.0.0
DEBUG: HandShake: Handshaking finished....
DEBUG: RTMP_Connect1, handshaked
DEBUG: Invoking connect
[RTMP]connect success!
DEBUG: HandleCtrl, received ctrl. type: 0, len: 6
DEBUG: HandleCtrl, Stream Begin 0
DEBUG: RTMP_ClientPacket, received: invoke 241 bytes
DEBUG: (object begin)
DEBUG: (object begin)
DEBUG: Property: <Name:             fmsVer, STRING:     FMS/3,0,1,123>
DEBUG: Property: <Name:       capabilities, NUMBER:     31.00>
DEBUG: Property: <Name:               mode, NUMBER:     1.00>
DEBUG: (object end)
DEBUG: (object begin)
DEBUG: Property: <Name:              level, STRING:     status>
DEBUG: Property: <Name:               code, STRING:     NetConnection.Connect.Success>
DEBUG: Property: <Name:        description, STRING:     Connection succeeded.>
DEBUG: Property: <Name:     objectEncoding, NUMBER:     0.00>
DEBUG: Property: <Name:               data, NUMBER:     0.00>
DEBUG: Property: <Name:            version, STRING:     3,5,1,516>
DEBUG: (object end)
DEBUG: (object end)
DEBUG: HandleInvoke, server invoking <_result>
DEBUG: HandleInvoke, received result for method call <connect>
DEBUG: Invoking releaseStream
DEBUG: Invoking FCPublish
DEBUG: Invoking createStream
DEBUG: RTMP_ClientPacket, received: invoke 20 bytes
DEBUG: (object begin)
DEBUG: Property: NULL
DEBUG: (object end)
DEBUG: HandleInvoke, server invoking <_result>
DEBUG: HandleInvoke, received result id 2 without matching request
DEBUG: RTMP_ClientPacket, received: invoke 24 bytes
DEBUG: (object begin)
DEBUG: Property: NULL
DEBUG: (object end)
DEBUG: HandleInvoke, server invoking <onFCPublish>
DEBUG: RTMP_ClientPacket, received: invoke 20 bytes
DEBUG: (object begin)
DEBUG: Property: NULL
DEBUG: (object end)
DEBUG: HandleInvoke, server invoking <_result>
DEBUG: HandleInvoke, received result id 3 without matching request
DEBUG: RTMP_ClientPacket, received: invoke 29 bytes
DEBUG: (object begin)
DEBUG: Property: NULL
DEBUG: (object end)
DEBUG: HandleInvoke, server invoking <_result>
DEBUG: HandleInvoke, received result for method call <createStream>
DEBUG: Invoking publish
DEBUG: RTMPSockBuf_Fill, recv returned -1. GetSockError(): 104 (Connection reset by peer)
ERROR: WriteN, RTMP send error 32 (59 bytes)
ERROR: Close Rtmp!

DEBUG: Invoking FCUnpublish
ERROR: RTMP_ReadPacket, failed to read RTMP packet header

Answer 1

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librtmp推流问题

你使用 librtmp 库进行推流，往虎牙直播推流正常，但往斗鱼直播推流时提示无效的推流码，推流失败。librtmp 打印如下：

[RTMP]Connect
DEBUG: Parsing...

可能的原因：

1.斗鱼直播的推流服务器可能需要特殊的头信息或参数，librtmp 库没有默认设置这些信息。 2.斗鱼直播的推流服务器可能需要特殊的加密或身份验证，librtmp 库没有默认设置这些信息。

解决方案：

1.检查斗鱼直播的推流文档，了解其推流服务器的需求和参数。 2.使用 librtmp 库提供的设置函数，设置头信息、参数、加密或身份验证等信息。 3.检查 librtmp 库的日志信息，了解推流失败的原因。

例如，你可以使用 rtmp_set_option 函数设置头信息和参数：

rtmp_set_option(rtmp, RTMP_OPTION_HEADERS, "Content-Type", "application/x-h264");
rtmp_set_option(rtmp, RTMP_OPTION_HEADERS, "Stream-Name", "your_stream_name");

你也可以使用 rtmp_set_option 函数设置加密或身份验证信息：

rtmp_set_option(rtmp, RTMP_OPTION_AUTH, "username", "your_username");
rtmp_set_option(rtmp, RTMP_OPTION_AUTH, "password", "your_password");

请注意，这只是一个示例代码，你需要根据斗鱼直播的推流文档和 librtmp 库的文档进行修改和设置。