Version 0.4.2 is not working with FreeSwitch

[IlyaRu]

Dear colleagues,

I made two similar examples:
http://cloud001.cxwebservice.com/test_030.html
http://cloud001.cxwebservice.com/test_042.html

The difference is in versions of JsSIP (0.3.0 vs 0.4.2) and event name (started vs confirmed). Connection established, but there is no sound in 0.4.2. What to do? :-)

[jmillan]

Hi @IlyaRu,

Beware that FF lately offers the 0.0.0.0 IP address in the SDP 'm=' line. c=IN IP4 0.0.0.0. So the server could correctly assume that the peer doesn't want to receive RTP nor RTCP as per RFC3264.

I guess you could get more help in the FreeSwitch mailing list.

Related FF issue: https://bugzilla.mozilla.org/show_bug.cgi?id=1072384

[jmillan]

Tested tryit.jssip.net in FF 33.0.2 as well as FF Nightly, both peers being FF. Media flows correctly.

Let me close this issue.

[IlyaRu]

Dear colleagues,

I am not using FireFox, I am using Google Chrome. Note, that JsSIP 0.3.0 works fine, but in 0.4.2 I hear silence.

Console logs of two calls are in attachments – signaling part is very similar. You can reproduce it easy:
http://cloud001.cxwebservice.com/test_030.html
http://cloud001.cxwebservice.com/test_042.html

[jmillan]

Ups, I read FreeSwitch and introduced FF in the equation myself.., let me reopen it.

[jmillan]

I can't see anything at a first glance, and doubt JsSIP has nothing to do. Is the only different thing in both scenarios the JsSIP library version?

[IlyaRu]

Yes, in these two html files there are two differences:

• version of library
• name of event started/confirmed.

Did you reproduced my problem in scenario test_042?

[jmillan]

Yes I did. No audio is reproduced.

[IlyaRu]

Can I help you to find the reason? May be some settings on my server?

[jmillan]

Sorry, I haven't had that time to take a look at this. It would be nice to know the exact commit after which the audio, in your scenario, is not reproduced anymore. If the only changing component is JsSIP.

0.3.0 works, 0.4.2 doesn't. Then taking a commit between those both versions and building JsSIP will tell whether the 'error' was introduced before or after such commit, and so on...

It's not funny, but It should in the end show the exact commit after which it stopped working.

[IlyaRu]

So, where can I find all intermediate versions between 0.3.0 and 0.4.2?

[jmillan]

This command shows you the commits between the tag 0.3.0 and the HEAD of the repository:
git log v0.3.0.. --pretty=format:"%h %s"

Going to the mid commit between the list and testing, then going upwards or backwards to the mid commit of the remaining list and so on, should take about 8 iterations to find the guilty.

[IlyaRu]

I found 3 commits with different behaviour:

13.11.2013 14:16 ed643f7 Merge pull request #149 from crocodilertc/issue-148
works fine

19.11.2013 12:11 62e8323 Fix #176. Update to the latest IceServer definition
connect failure (Uncaught TypeError: Failed to construct 'RTCPeerConnection': Malformed URL )

14.07.2014 14:32 91bef59 Merge develop branch in master.
silence

There is a lot of differences between 13.11.2013 and 14.07.2014. Have you any ideas?

[ibc]

@IlyaRu can you provide a public FS WebSocket URI in which the problem can be reproduced?

[IlyaRu]

Dear colleagues,

I have a very interesting observation. I have created 2 accounts on my FreeSwitch: 777 and 888. The difference is only in the sequence of actions:
777: answer call -> start recording -> play sound file
888: start recording -> answer call -> play sound file

777 works fine with JsSIP 0.3.0 and 0.4.2
888 works with 0.3.0 and has no sound with 0.4.2.

I will do some investigations to understand the difference.

WS parameters (will be valid for some days):

    'ws_servers': 'ws://148.251.182.78:5066',

    'uri': 'sip:[email protected]',
    'password': '888',

    'uri': 'sip:[email protected]',
    'password': '777',

[IlyaRu]

Colleagues,

the difference of 777 and 888 is in SIP messages flow. 777 answers 200 OK immediately. 888 answers 183 Progress, then OK.

So, in JsSIP 0.4.2 processing of message 183 seems to be broken.

[ibc]

We don't have a FS installation. Any chance to test one provided by you?

[IlyaRu]

I provided you WS accounts on my FS:

'ws_servers': 'ws://148.251.182.78:5066',

'uri': 'sip:[email protected]',
'password': '888',

'uri': 'sip:[email protected]',
'password': '777',

777 works fine with JsSIP 0.3.0 and 0.4.2
888 works with 0.3.0 and has no sound with 0.4.2.

Besides, you can try my test HTML pages:
http://cloud001.cxwebservice.com/test_030.html
http://cloud001.cxwebservice.com/test_042.html

Of course, I can test new version of JsSIP when it will be available.

[wilshire]

Is there an update on this issue? I am trying to upgrade to v0.5.0 from 0.3.0 and encountered the same issue. I can say that it looks to me like the STUN negotiation seems to go south and settles on random ports. Anything I can do?

[ibc]

JsSIP does not deal with STUN, that's browser's business.
I will check this when I have some time, but I usually don't have time to test JsSIP in different browsers with different media servers. If you can reproduce this issue in your environment, then please try to find out the issue. Pull requests are welcome.

[jmillan]

What about newer versions?

Anyone else out there using FreeSwitch?

[ghost]

I have tested 0.5 with Freeswitch... It worked after some minor modifications.

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José Luis Millán [email protected] hat am 28. Januar 2015 um 21:36
geschrieben:

What about newer versions?

Anyone else out there using FreeSwitch?

---

Reply to this email directly or view it on GitHub:
#262 (comment)

[ibc]

Sorry @hoene, that is not a solution at all.

[fxalgrain]

Hi everybody,

I m using JsSIP last version with a Freeswitch server too.

I have the same issue, works fine on Firefox and no sound on Chrome.