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gst1-plugins-bad: FTBFS #597

Closed sbyx closed 9 years ago

sbyx commented 9 years ago

gstsrtp.c:183:29: error: 'AES_ICM' undeclared (first use in this function) policy->cipher_type = AES_ICM;

http://buildbot.openwrt.org:8010/broken_packages/ar71xx.mikrotik/gst1-plugins-bad/compile.txt @MikePetullo @thess

thess commented 9 years ago

The problem only shows up if libsrtp from the telephony repo is built. Normally, libgstsrtp is not selected to be built, but configure finds it anyway. Fixed by explicitly disabling srtp. 0cbca8677db248ef9295e7dbf6d99b2766801f09

The version of libsrtp in the telephony repo is ancient. A more current version from https://github.com/cisco/libsrtp should be imported - @jslachta ?

jslachta commented 9 years ago

@thess Thank you for mentioning, I did not know that the libsrtp moved to github. I am going to fix it as soon as possible.

jslachta commented 9 years ago

@thess May I ask you for the test whether srtp works? If so, would you be so kind to revert this change? I updated srtp package a second ago.

thess commented 9 years ago

Done...

enigmagroup commented 9 years ago

@jslachta res_srtp.so isn't loading anymore. I had to 'blacklist' it in asterisk's module.conf, and when loading it manually, it says:

module load res_srtp.so Unable to load module res_srtp.so Command 'module load res_srtp.so' failed. [Dec 10 12:44:47] WARNING[3262]: res_srtp.c:563 res_srtp_init: Failed to initialize libsrtp

jslachta commented 9 years ago

@enigmagroup Please, open a new issue for this. I will check it today and fix it as soon as possible. It seems that libsrtp is quite new for the asterisk package.

sbyx commented 9 years ago

@jslachta btw. what do you think about putting the telephony feed on github as well, either in a separate repository or by merging it with this feed?

that might make tracking such cross-feed issues a bit easier, but well it's up to you in the end ;)

enigmagroup commented 9 years ago

@jslachta ok thanks for your answer, I will.

jslachta commented 9 years ago

@sbyx I am all for it. Since there are repositories for management and routing, I would keep it in separate context in the same way. I just did not know who should I ask for the repository transfer, since I do not know who maintains the repository itself on git.openwrt.org. I am just a content provider for (almost) the whole repository.

sbyx commented 9 years ago

@jslachta I created and imported a repository here under https://github.com/openwrt/telephony and will tell @nbd168 who maintains git.openwrt.org to make the repo there a mirror once you are comfortable with it.

thess commented 9 years ago

@sbyx - Don't forget to update feeds.conf for this change. And, please, refrain from using this thread to discuss the telephony repo. ;)

jslachta commented 9 years ago

@thess I am sorry for reopening this issue.

Since the libsrtp update breaks the whole SRTP channel functionality in Asterisk packages, I decided to temporarily downgrade the libsrtp Makefile to the previous version. I tried to find the problem, but I was currently unable to resolve it. Because of the upcoming CC which is quite close I would like to keep the main things working. When it is going to be fixed, I am going to readd the plugin back.

Would you be so kind to temporarily revert the last commit which adds back the srtp functionality?

I am so sorry for any inconveniences. Thank you so much.

thess commented 9 years ago

Done - for now. be72d3b10c144acc7cf1739b77df150c96f8ee4e