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| 01:50.10 | *** topic/#asterisk-dev is Asterisk Development Discussion -=- http://www.asterisk.org/developers -=- Tier 2 and 3.14159265 support is in #asterisk -=- Check out our blog! blogs.asterisk.org -=- Follow on Twitter at @AsteriskDev |
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| 10:11.43 | persiever | Can I get source code on how to stream audio file (wav)and video file (h263) in a single port ? |
| 10:12.53 | file | the only source code in Asterisk that does such a thing is bundle, otherwise it's something you have to define and write (for example if it's not RTP then how are the packets differentiated) |
| 10:14.17 | persiever | what does the bundle refers to? Does it means mixing audio and video. Or something like multiplexing? |
| 10:14.44 | file | it's a specification for SIP and SDP to bundle media streams over the same port. |
| 10:15.16 | file | but it's really not that hard to do such a thing, it just depends on how they are represented and differentiated |
| 10:15.19 | file | which is specific to your usage |
| 10:15.48 | tsearle | doesn't chan_iax also multiplex on a single port too? |
| 10:16.01 | file | tsearle: true true |
| 10:18.17 | persiever | ok, As I have configured chan_sip in sip.conf file for two stream (gsm,h263). In invite I can see two port; one for audio RTP/AVP and other for video RTP/AVP. Thus if both port are provided in SDP Session progress from other party, then video call progress. But current scenario is, I am getting a Single port, where I have to send both audio and video |
| 10:18.36 | file | chan_sip does not support such a thing. |
| 10:18.47 | persiever | So how I will configure chan_iax for that. What changes need to done in code level? |
| 10:19.02 | file | chan_iax2 is a completely separate protocol that isn't SIP. |
| 10:25.14 | persiever | from where I can get the source that will help me to write code for streaming audio & video in same RTP port? |
| 10:25.22 | persiever | any suggestion? |
| 10:26.52 | file | the res_rtp_asterisk.c module supports bundle as I stated which can place audio and video on the same RTP port |
| 10:27.11 | file | but "in same RTP port" doesn't mean much, as I stated before, without stating HOW that is done |
| 10:27.41 | file | cause you can't just put it on the same RTP port, it has to be defined how it will work |
| 10:29.12 | persiever | i can define two codecs like amr, h263 with same payload in res_rtp_asterisk.c and probably it will send that in SDP headers. Doesn't it will conflict? |
| 10:29.51 | file | they could not be negotiated at the same payload, depending on the code it may break things |
| 10:31.42 | persiever | I think solution will be a new codec. I have name for it CLEARMODE/8000 codec as defined by RFC4040. As this codec provides only packetization and depackatization. We can send data in it |
| 10:32.24 | file | I have no knowledge of the thing you are ultimately trying to implement, so I do not know |
| 10:34.33 | persiever | ok, I want to know how to read the file (gsm +h263) and how they stream to dynamically assigned port setup by sdp negotiation. So that I can write code for required functionality |
| 10:35.24 | file | there are modules to already read such files, and both chan_sip and chan_pjsip use the RTP API to set ports up, and they do their SDP negotiation |
| 10:36.42 | file | the easiest way is to look at the existing code which utilizes them |
| 10:38.25 | persiever | can you suggest a source file name from where I can look into and utilize that? Probably I want to look whole code that is involve in RTP stream. |
| 10:38.44 | file | res_rtp_asterisk.c, res_pjsip_sdp_rtp.c |
| 10:39.20 | file | rtp_engine.c |
| 10:39.40 | file | this may ultimately be beyond your skill set, though |
| 10:40.33 | persiever | ok thanks. |
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| 12:45.59 | pchero_work | Hi, I have a question regarding sip session supplement. I registered callback function for sip incoming response. But when I tried to access to the rdata->msg_info.msg , it's NULL everytime. Did I miss something? |
| 12:47.47 | file | is it a supplement or a module? did you look at another module/user and see what they do? |
| 12:48.45 | pchero_work | Ah, I'm making new module using supplement. I referenced this tutorial. https://wiki.asterisk.org/wiki/display/AST/Writing+a+SIP+Session+Supplement |
| 12:49.24 | pchero_work | example in the tutorial was fine. but rdata->msg_info.msg is always NULL.. |
| 12:50.24 | file | I wouldn't expect it to be NULL, how are you building the module? dropping it into Asterisk? |
| 12:50.59 | pchero_work | Yes, I loaded it in the Asterisk using "moudle load â¦" cli command. |
| 12:51.13 | file | right, but how did you build it? |
| 12:52.22 | pchero_work | Compiled it with gcc� I made some other asterisk-modules and submitted Asterisk's bug fixes before.. |
| 12:52.39 | file | externally from the Asterisk source code, or in the tree? |
| 12:52.54 | pchero_work | Ah, externally. |
| 12:52.59 | file | it's important because if not done correctly, your module could be compiled against a different PJSIP - resulting in pjsip_rx_data having a different layout |
| 12:53.11 | pchero_work | Ah.. |
| 12:53.11 | file | and thus it would be NULL because the struct wouldn't be the same |
| 12:53.19 | pchero_work | You're right. |
| 12:53.34 | pchero_work | Yes, I installed the other pjsip library for this. |
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| 12:54.03 | file | that's likely why |
| 12:54.12 | pchero_work | file: Thanks. I'll test it again with Asterisk's pjsip library. |
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| 13:34.09 | pchero_work | file: Thank you. :) You're right. Btw, it there any easy options to install the Asterisk's third-party's into local include/library path? |
| 13:34.24 | file | I do not know, gtjoseph might |
| 13:35.27 | gtjoseph | pchero_work: you'd have to go into the source directory and do a ./configure and make install manually |
| 13:36.08 | pchero_work | gtjoseph: Ah.. no easy way. :'( Thank you! Now I'm clear. :D |
| 13:46.37 | file | coreyfarrell: poke |
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| 13:55.09 | coreyfarrell | file: about installing third-party to a system path that's not what it's designed for. pjproject is configured with --prefix=/opt/pjproject so you could possibly edit third-party/pjproject/Makefile.rules to repoint prefix. that way you could possibly make asterisk download/patch/configure/build pjproject, then go into third-party/pjproject/source and 'make install' (this is completely untested). |
| 13:55.31 | file | hmm? |
| 13:56.17 | coreyfarrell | sorry were you not poking about pchero_work's question? |
| 13:57.01 | coreyfarrell | I probably should have tagged pchero_work instead of you |
| 14:01.00 | file | I was not! |
| 14:01.07 | file | I was wondering if you ran the ptime tests |
| 14:07.32 | coreyfarrell | file: no I didn't even download it. my comments on that review were just from looking at the patch in gerrit. I assume the tests will leak as is because they perform pjsip INVITE's. |
| 14:08.54 | file | ack |
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| 14:33.51 | tsearle | hey, I have a question ref your jenkins |
| 14:33.56 | tsearle | when the build fails with |
| 14:33.59 | tsearle | Test Result (72 failures / +3) |
| 14:34.11 | tsearle | do I need to look at all 72 failures, or just the +3? |
| 14:34.23 | tsearle | and if the latter, how can I tell which are the 3 new failures? |
| 14:34.37 | gtjoseph | in normal gates, there should be no more than a handful of failures |
| 14:34.44 | gtjoseph | 72 is a LOT |
| 14:34.55 | tsearle | https://jenkins2.asterisk.org/job/asterisk-gatetests/job/13/226/#showFailuresLink |
| 14:35.01 | file | most of those failures are because of your change, I cherry picked a few to look at and they were because of the NOTIFY going to the wrong place |
| 14:35.08 | file | and the +3 may have been because I regated |
| 14:35.20 | tsearle | ok, so I need to look at all 72 |
| 14:36.24 | bford | or just look at a couple and maybe you can figure out the issue from there - 72 would be a nightmare |
| 14:38.49 | tsearle | hrm... how do I get the asterisk logs for this testsuite run? downloaded the atifact, but unline our jenkins, there isn't a logs/path/to/faild/test directory |
| 14:39.34 | file | you can browse them in the UI if you want |
| 14:39.37 | tsearle | ideally I would llke the sipp log, but you disable sipp-loging in your test suite |
| 14:40.06 | file | I just looked at the Asterisk logs |
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| 15:15.10 | tsearle | if I revise the patch in gerrit, will your jenkins re-run the testsuite, or will that only happen on a +2? |
| 15:15.57 | file | it will be re-reviewed, and then once another +2 is done it'll be regated |
| 15:16.39 | tsearle | ok, so using your jenkins testsuite isn't really an option |
| 15:17.01 | file | I don't understand the question |
| 15:17.14 | file | oh |
| 15:17.19 | file | it only happens on a +2 |
| 15:17.27 | tsearle | the testsuite we run doesn't have any of the 72 tests that are currently failing |
| 15:17.52 | tsearle | I can try copying a couple over, but probably not all 72, heh |
| 15:47.01 | pchero_work | coreyfarrell: file: Thanks. I've just checked your messages. :) |
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