sip

Failed to create SipSession; network unavailable and only reboot helps

由 别说谁变了你拦得住时间么 提交于 2019-12-21 20:11:11
问题 I am developing a simple application that allows user to initiate call using sip protocol. The problem is inability to create SipSession in some cases (e.g. deleting application with active sip-session, and installing it again). In this case, I am getting error: android.net.sip.SipException: Failed to create SipSession; network unavailable? And it works only after physical device reboot. My Sip class: public class SipDataManager { private Context context; private SipManager sipManager;

VoLTE SIP代码意义及流程图解

由 耗尽温柔 提交于 2019-12-21 07:17:43
VoLTE SIP代码意义及流程图解 本帖最后由 ufooo 于 2016-2-25 10:55 编辑 2016-2-25 10:45 上传 下载附件 (65.13 KB) </ignore_js_op> 一 VOLTE SIP代码意义 SIP应答消息状态码与功能 类型 状态码 状态说明 临时应答(1XX) 100 Trying 正在处理中 180 Ringing 振铃 181 call being forwarder 呼叫正在前向 182 queue 排队 181* session progress 会话进行 会话成功(2XX) 200 OK 会话成功 重定向(3XX) 300 multiple 多重选择 301 moved permanently 永久移动 302 moved temporaily 临时移动 305 use proxy 用户代理 380 alternative service 替代服务 请求失败(4XX) 400 bad request 错误请求 401unauthorized 未授权 402 payment required 付费要求 403 forbidden 禁止 404 not found 未发现 405 method no allowed 方法不允许 406 not acceptable 不可接受 407 proxy authentication

SIP协议整理

由 旧街凉风 提交于 2019-12-21 07:15:56
　　本文记录开发、实现IMS项目时，整理的SIP协议基础知识；若有侵权，请告之。 SIP协议 1. SIP 协议简介 SIP是一个应用层的控制协议，可以用来建立、修改、和终止多媒体会话（或者会议）例如Internet电话。SIP也可以邀请参与者参加已经存在的会话，比如多方会议。媒体可以在一个已经存在的会话中方便的增加（或者删除）。SIP显示的支持名字映射和重定向服务，这个用于支持个人移动业务－用户可以使用一个唯一的外部标志而不用关系他们的实际网络地点。SIP在建立和维持终止多媒体会话协议上，支持5个方面： 用户定位 ：检查终端用户的位置，用于通讯。 用户有效性 ：检查用户参与会话的意愿程度。 用户能力 ：检查媒体和媒体的参数。 建立会话 ：”ringing”,建立会话参数在呼叫方和被叫方。 会话管理 ：包括发送和终止会话，修改会话参数，激活服务等等。 SIP不是一个垂直集成的通讯系统。SIP可能叫做是一个部件更合适，它可以用作其他IETF协议的一个部分，用来构造完整的多媒体架构。比如，这些架构将会包含实时数据传输协议（RTP）（RFC1889）用来传输实时的数据并且提供QoS反馈，实时流协议（RSTP）(RFC2326)用于控制流媒体的的传输，媒体网关控制协议（MEGACO）(RFC3015)用来控制到公共电话交换网（PSTN）的网关，还有会话描述协议（SDP）（RFC2327

How to make live voice phone call using Twilio instead of just playing an MP3 when call is answered?

由 风流意气都作罢 提交于 2019-12-21 06:08:17
问题 To call phone number from notebook through Twilio I created ASP.NET-MVC 5.2 application. I can call a number and answer the phone but I don't know how to achieve live voice(to be able to talk) connection instead of just playing music. I created an action method inside HomeController : public ActionResult Call(string to) { client = new TwilioRestClient(Settings.AccountSid, Settings.AuthToken); var result = client.InitiateOutboundCall(Settings.TwilioNumber, to, "http://twimlets.com/message

How to make live voice phone call using Twilio instead of just playing an MP3 when call is answered?

由 一笑奈何 提交于 2019-12-21 06:07:32
问题 To call phone number from notebook through Twilio I created ASP.NET-MVC 5.2 application. I can call a number and answer the phone but I don't know how to achieve live voice(to be able to talk) connection instead of just playing music. I created an action method inside HomeController : public ActionResult Call(string to) { client = new TwilioRestClient(Settings.AccountSid, Settings.AuthToken); var result = client.InitiateOutboundCall(Settings.TwilioNumber, to, "http://twimlets.com/message

Incoming calls with SIP and WebRTC [closed]

由 时光毁灭记忆、已成空白 提交于 2019-12-21 04:12:23
问题 Closed . This question needs to be more focused. It is not currently accepting answers. Want to improve this question? Update the question so it focuses on one problem only by editing this post. Closed 3 months ago . I'm looking into implementing a browser-based VOIP solution that uses SIP and WebRTC and that connects to the PTSN. Basically, users give me their SIP credentials and I use WebRTC to acccess their microphone and speakers. On the page, I plan to run an SIP client. How do I

sipdroid - Another incoming call is not display while one is continue

由 元气小坏坏 提交于 2019-12-20 19:48:12
问题 I have installed SIPDROID . I made three account at sip2sip. After that I have configured application using one of that account. Application works fine. Question is: If I make multiple call to the same phone using SIP is there anyway to handle all calls at a same time? Another incoming call is not displaying on mobile screen while first call is continue. The first call getting picked up but what about the second call which is coming during first call. Even second call is not displaying on the

Identify h264 profile and level from profile-level-id in sdp?

由 Deadly 提交于 2019-12-20 14:43:10
问题 In a SIP video call, the receiver of the video stream respond with the capabilities of its decoder. The parameter which defines that is the profile-level-id . Here is an example value of the profile-level-id parameter: 428014 RFC 6184 defined that as A base16 [7] (hexadecimal) representation of the following three bytes in the sequence parameter set NAL unit is specified in 1: 1) profile_idc, 2) a byte herein referred to as profile-iop, composed of the values of constraint_set0_flag,

Peer-to-peer SIP call with Android SIP Stack?

由 久未见 提交于 2019-12-20 10:46:55
问题 I have been looking for a way to set up the Android SIP stack to be able to establish a SIP call between two devices on the same network, in an ad-hoc manner. i.e without REGISTERing to a SIP server. I have not been able to get this to work, as the SIP Demo includes server registration, and I cannot get it to make or receive a call without this step. I am not even sure if this is supposed to be possible. The little mention of this I have been able to find is conflicting (some say it can be

sip协议的超时机制

由 我的梦境 提交于 2019-12-20 10:24:51
0、前言： 在讲解sip协议的超时机制之前，先简单地介绍一下sip协议中的message、dialog、session 和 transaction （1）M essages(消息) 消息是在服务器和客户端之间交换的独立文本，有两种类型的消息,分别是请求(Requests)和响应(Responses)。 （2）Dialog(对话) 对话是两个UAs(user agent) 之间持续一段时间的端到端(peer-to-peer)的SIP 关系.。一个对话由一个Call-ID、一个local tag 和 一个remote tag来标识，对话过去也叫做 "call leg"。dialog的建立是收到UAS的响应（To tag）时开始建立的。收到180响应时建立dialog叫做早期对话（early dialog），收到2XX的应答开始才是真正的dialog建立。 当UA发送初始INVITE请求后，只有接收到失败响应才有可能建立DIALOG。通过Callid、From域中的tag参数和To域中的tag参数来唯一标识DIALOG。 From域中的参数由主叫添加，To域中的参数由被叫添加。 （3）Transaction(事务) 事务发生于客户端和服务器端之间，包含从客户端发出请求给服务器，到服务器响应给客户端的最终消息(non-1xx message)之间的所有消息.。如果请求是一个"Invite
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