[ERROR]: chan_sip.c compiling issue, svn undeclared

Asterisk Asterisk Support
1

Hello,

I am a rookie to asterisk. I am using Asterisk 11.18 and was trying to run WebRTC using Asterisk.
I was following instructions from here.

https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5

After the following step “Build Asterisk with support for WebRTC”, I got this error. Could anyone help me fix this error?

error: chan_sip.c: In function ‘add_ice_to_sdp’: chan_sip.c:12575:53: error: ‘generation’ undeclared (first use in this function) chan_sip.c:12575:53: note: each undeclared identifier is reported only once for each function it appears in chan_sip.c:12575:65: error: ‘svn’ undeclared (first use in this function) make[1]: *** [chan_sip.o] Error 1 make: *** [channels] Error 2

This is the line in chan_sip.c where I am getting the error:

2

That code does not appear to be in Asterisk 11.18.0 - where did you get it from? Did you add it?

3

This is the code from the file chan_sip.c

I am to connect the calls but no audio flows. I set rtp debug on and figured out it was due to no ICE support. Eventually, it is because chan_sip.c is not enabled. When I am trying to enable the file by using menuselect, it is giving me above error.

The code is from asterisk 11.18.0 only. I did not add it. It is at line number 12575 in chan_sip.c

4

Line 12575 of chan_sip.c in 11.18.0 is:

Which does not match what you have provided.

5

I dont know where that line came from. It was right before the line which you have posted. I commented the line and it didn’t give any further compilation error. However, does that have any impact on the ICE support enabling? I am now having issue with icesupport. Can’t see “via ice” when doing rtp debug. And I feel its all related to chan_sip.c

6

It should not have an impact. To look at ICE issues you’d need to provide a full console output including SIP traffic.