go-sip-ua
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cloudwebrtc
Go SIP UA library for client/b2bua
I'm trying to use your pkg, and I find myself facing multiple worries. First of all, in the [endpoint.go](https://github.com/cloudwebrtc/go-sip-ua/blob/master/pkg/endpoint/endpoint.go) file, the _GetNetworkInfo_ function doesn't take the value Endpoint.ip It makes...
Hello, I wanted to try your library but I never managed to get the audio streams back. More concretely, I use freeswitch like ipbx, I can connect myself, receive the...
## Reason `auth.nc` with the int type creates **overflow int** once building for `arm` architecture ( in my case Raspberry Pi 3) ## Solution Typecasting `auth.nc` for the `if` clause.
 如图 这个一直都是返回 未找到 不管是哪一方挂电话 都没有满足这个条件 所以里面的Terminated一直都没有执行到。
Currently, log levels of SIP stack, transport and transaction layers are set to debug. There is a way to change SIP stack log level but there is no way to...
Problem: Can not connenct to SIP server when client behind a NAT network. Request Log: ``` REGISTER sip:[email protected]:7060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.187:50066;branch=z9hG4bK.iJM9FQ5m5ZRW0ZJZTAvNGlTyN9JPA6Nl ...... ``` What's expected: ``` REGISTER sip:[email protected]:7060 SIP/2.0...
In current version of code, the "Via" in register request looks like: `Via: SIP/2.0/UDP 192.168.1.100:50001;branch=z9hG4bK.iJM9FQ5m5ZRW0ZJZTAvNGlTyN9JPA6Nl` In a common VoIP software, the "Via" in register request looks like: `Via: SIP/2.0/UDP 192.168.1.100:50001;rport;branch=z9hG4bK.iJM9FQ5m5ZRW0ZJZTAvNGlTyN9JPA6Nl`...
Hi! Thanks for this useful project. I am facing the problem of not being able to handle session termination when receiving a BYE request. I found that the CallId +...
SIP中不能仅仅用callID和fromTag来获取session，在你这种设计方法里，可能需要先用一个map来判断方向，根据方向来使用fromTag或者toTag来获取session  
