sip

pjsua custom sdp

由 ≡放荡痞女 提交于 2019-12-08 06:24:00
问题 I'm using pjsua to create simple SIP UA. I need to insert custom body (SDP) in outgoing INVITE requests. As far as I know, pjsua does not support custom SDP, and I need to use more low-level API to do it. But for now I have to find any rude trick/hack/hotfix to insert custom SDP in pjsua-generated INVITE. So that's the question: how could I do it? 回答1: The seemingly simple solution is to modify the pjsua source. The key is to create a pjmedia_session out of your custom SDP on code paths of

How to send RTPPackets after SIP call Invite request method?

由 不想你离开。 提交于 2019-12-08 01:59:38
问题 I am developing an application based on VOIP(SIP) . I am able to send Invite and Invite-Ack between two emulators but now i would like to send RTPpacket after Invite-ack message in SIP call flow. Is it possible to send like that. If yes, please can anyone help me regarding this. After that i m going to integrate this for both send and receive part of SIP video call. Any help would be appreciated lot!!! Thanks in Advance!!!!! 回答1: i think i will have to look at the RTP implementation of

How to detemine which network interface (ip address) will be used to send a packet to a specific ip address?

由 放肆的年华 提交于 2019-12-08 01:39:20
问题 I'm writing a SIP stack, and I need to insert an ip address in the message. This address needs to be the one used for sending the message. I know the destination IP and need to determine the NIC (its address) that will be used to send the message.... 回答1: To expand a bit on Remy Lebeau's comment, GetBestInterfaceEx() is your best bet, if you're on Windows XP or newer. That will work for both IPv4 and IPv6 addresses. GetBestInterface/GetBestInterfaceEx return the index (call it IDX) of the

SIP servlets, chatserver

由 僤鯓⒐⒋嵵緔 提交于 2019-12-08 01:36:25
I'm trying to get a SIP servlet chat server working, together with the textclient found here . When I use 2 clients to send messages to eachother (peer to peer), everything goes well. But when I use one or more clients together with my server, I have to wait exactly 32 seconds before the server picks up any new messages in the doMessage() method. I'm using Netbeans together with Sailfin as my SIP server. Is there some kind of limitation or configurable delay or timeout between requests or responses in Sailfin I'm looking over? I can post the server code, if needed. Thanks Here is the code of

Android-ready JAIN-SIP library?

由 流过昼夜 提交于 2019-12-07 10:27:17
问题 Greets! I am developing (trying to develop) a VoIP SIP application for Android, and after two weeks of bickering with mjsip, pjsip and the sdk's libraries, I have settled on JAIN-SIP. The libraries look great, in theory. No need to learn any android native code, lots of documentation (yet not enough, since I'm here), etc. My first attempt was using the Android SDK's SIP libs (yes, I know they're based on JSIP), and it failed after the SipManager.open() refused to open the profile for

W/AudioGroup﹕ device loop timeout

由 一世执手 提交于 2019-12-07 09:37:59
问题 i'm trying to do a SIP call in my android. Sound works great on the first call, but when this call is ended and i start a second call, sound is interrupted and i get following warning (multiple lines per second): W/AudioGroup﹕ device loop timeout` I start the call with: mSipManager.makeAudioCall( getAccount().getLocalSipProfile(), getAccount().getPeerSipProfile(), new SipCallListener(), 0 ); Contents of call listener: @Override public void onCallEstablished(SipAudioCall call) { call

Why does the native SIP stack included in Android 2.3 does not work over 3g?

由 守給你的承諾、 提交于 2019-12-07 07:20:53
问题 I was wondering why does the native SIP stack included in the Android framework(since 2.3) does not work over 3g? Could it have something to do with any laws or restrictions google may have with his partners? And furthermore, does anybody know if there is any plans to remove that restriction ? Thx 回答1: In GingerBread, SipManager is set to work only on wifi. `<bool name="config_sip_wifi_only">true</bool>` But from 4.0 onwards, this config has been changed to false So Ideally native sip stack

SIP over websockets to true SIP

由 ￣綄美尐妖づ 提交于 2019-12-07 06:20:10
问题 I'm trying to implement a sip server for connecting to from an HTML sip client(made using sipml5). During my research into doing this I've come across sip over web-sockets which might be useful to me, however, I am unsure if a user agent connecting through sip over web-sockets to a compatible server would then be able to successfully make a call to some one using an incompatible server(i.e. calling from SIP over web-sockets to true SIP). I know webrtc2sip can be used for connecting to legacy

In SIP, why the Contact header field MUST be present in the Invite request

由 試著忘記壹切 提交于 2019-12-06 23:09:45
问题 Usually, the Contact header field in the Invite request is useless. For example, the UAC and the UAS are in different LANs. The Contact field may be: INVITE sip:bob@sipprovider SIP/2.0 Contact: Alice<alice@192.168.1.10> ..... There is no use of the Contact field while we can still build a dialog. Then, why the Contact header field is mandatory？ 回答1: The contact field contains the address at which the callee can reach the caller for future requests. For example, it's necessary so that the

Is it possible to use MJSIP api with Blackberry?

由 亡梦爱人 提交于 2019-12-06 15:29:42
I am trying to develop an VOIP application for blackberry,after a long surf i came to know about mjsip api.But i have a doubt that is it possible to use this api with blackberry development to create VOIP application.Please anyone knows the answer help me. Thanks, it's a nice project you found there! There is an J2ME version MjSipME , and the only thing I can say now for sure is that it compiles with Blackberry without any errors. UPDATE You right, there are missunderstanding with packages/folders structure. Steps to compile: download mjsip2me_1.6.zip create blackberry project (I've used
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