Ecosyste.ms: Timeline
Browse the timeline of events for every public repo on GitHub. Data updated hourly from GH Archive.
sipsorcery pushed 3 commits to gh-pages sipsorcery-org/sipsorcery
sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
Sorry for the delayed response. DTMF does not get a lot of love these days. More than happy to take a PR if you're still interested. Your approach sounds sensible.
sipsorcery closed an issue on sipsorcery-org/sipsorcery
Transmitting in-band RTP events and audio packets on a media stream
Hi, Thanks for all your work on sipsorcery. **Summary** I noticed that SendDtmfEvent was moved out of RTPSession.cs and into AudioStream.cs since v6.0.8. With this change, I think it has b...sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
There have been a number of improvements to the WebRTC peer negotiation that may have fixed this issue. If you're still interested and it's still a problem please feel free to re-open this issue,
sipsorcery closed an issue on sipsorcery-org/sipsorcery
Race condition - Webrtc Ice connection fails sometimes
I use Sipsorcery for connecting to a remote Webrtc peer (mobile Chrome browser) at the same WIFI network. The connection with Sipsorcery sometime fails (Race condition) if I enable VPN client at...sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
If you use the [Call method](https://github.com/sipsorcery-org/sipsorcery/blob/de91be67f8c79d6fd67bb5b18a7f9f33bd9ba6f5/src/app/SIPUserAgents/SIPUserAgent.cs#L485) that takes a SIPCallDescriptor yo...
sipsorcery closed an issue on sipsorcery-org/sipsorcery
Cannot pass message content while initiating sip call
As per RFC we are allowed to send different headers and body with SIP invite. In this case we can pass content type as multipart/mixed and body consisting of multiple contents separated by boundary...sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
I realise this answer is very later but just in case it's still relevant. Is it possilbe the TCP connection is still in progress rather than timed out? The SIPTCPChannel does not add a reocrd...
sipsorcery closed an issue on sipsorcery-org/sipsorcery
SIPTCPChannel. TCP transfer fails if the server was not available at the time of first use.
SIPTCPChannel is using port for listened, but don't add it connection if server not answer (not available). Next send can't create connection because the port busy. ![image](https://user-images.g...sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
I realise this answer is somewhat late. SIP and STUN are very different protocols. The way they interact is if your SIP device is behind a NAT you will often use a STUN client to determine what ...
sipsorcery closed an issue on sipsorcery-org/sipsorcery
How to configure STUN Client?
Hello, I am new to this SipSorcery and this STUN stuff. Sorry in advance if I miss interpret something. I am trying to connect to STUN server on the sip.us "https://support.sip.us/hc/en-us/artic...sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
You can use SIP subscriptions to get NOTIFY requests about events on remote SIP user agents. It will depend on the remote device supporting subscriptions but it's a failry common scenario.
sipsorcery closed an issue on sipsorcery-org/sipsorcery
Call transfer queue
I'm trying to create a queue for call transfers. The problem is that when we do a blind or attended transfer, the SIP server sends us a BYE on the original call so we have no means of knowing when ...sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
Default ordered to true in #1222.
sipsorcery closed an issue on sipsorcery-org/sipsorcery
`RTCDataChannel.ordered` is wrong
I understand that only ordered reliable channels are implemented, but it is very wrong that `RTCDataChannel` for newly opened channels reports `ordered` as `false` even though it is definitely `true`.sipsorcery pushed 1 commit to master sipsorcery-org/sipsorcery
- DEfault the webrtc datachannel ordered property to true. (#1222) de91be6
sipsorcery opened a pull request on sipsorcery-org/sipsorcery
Default the webrtc datachannel ordered property to true.
sipsorcery pushed 4 commits to gh-pages sipsorcery-org/sipsorcery
sipsorcery created a branch on sipsorcery-org/sipsorcery
datachannel-default-ordered - A WebRTC, SIP and VoIP library for C# and .NET. Designed for real-time communications apps.
sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
If you think it's being caused by the NAudio settings the place to start is in the [SIPSorceryMedia.Windows code](https://github.com/sipsorcery-org/SIPSorceryMedia.Windows/blob/5b2367169cdffb1d870f...
sipsorcery closed an issue on sipsorcery-org/sipsorcery
High latency in one direction only when connected to conference call
We have a phone system running on Asterisk, with both a SipSorcery instance and a Cisco IP phone connecting to a conference call. Audio from the softphone to the handset is sent with a minimal l...sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
The whole SDP & media re-negotiation logic was never implemented (it was hard enough to get things working with a single audio and video track). It's a sizeable job.
sipsorcery created a comment on an issue on sipsorcery-org/sipsorcery
I agree with @SteveAyre and did notice the same thing. It's wrong to add non-existent attributes when parsing an SDP payload. For SDP being produced it's also a very crude mechanism that is adding ...
sipsorcery pushed 1 commit to master sipsorcery-org/sipsorcery
- Comment out the wholesale addition of the feedback attribute to every media attribute. (#1221) c0ed0e8
sipsorcery created a branch on sipsorcery-org/sipsorcery
sdp-remove-fb-attr - A WebRTC, SIP and VoIP library for C# and .NET. Designed for real-time communications apps.