[Freeswitch-dev] mod_sofia hold/unhold issue correction problems
Dmitriy Borisov
borik.internet at gmail.com
Thu Jul 27 11:03:59 UTC 2017
Hi Mike,
Thank you for your reply.
I think the issue is within Freeswitch since FS is not putting the other
leg in the active state again,
We have the following scenario:
1. Outbound call from A to B over Freeswitch: A ==>> Freeswitch ==>> B
2. A puts the call on hold using inactive flag within the SDP header, FS
starts playing musiconhold towards B.
3. A sends a reinvite without a SDP header to FS (which should retrieves
the call from the hold scenario, since we’ve set sip_unhold_nosdp=true
4. FS keeps playing musiconhold towards B, while it should have put all
legs back in active state in my opinion.
It looks like this issue is related to JIRA FS-9983
Can you please confirm my conclusion and let me know in which part of the
code it’s possible to change this behaviour?
If you’re able to fix it I would be very happy as well of course.
Best regards,
пн, 24 июл. 2017 г. в 23:49, Dmitriy Borisov <borik.internet at gmail.com>:
> Hello, Michael!
>
> Problem not on the client side. On the client side call is active, not on
> hold. But the channel state in FreeSWITCH is holded. To enable voice I need
> to trigger hold twice (put on hold and take off hold after that)
>
> пн, 24 июля 2017 г., 19:19 Michael Jerris <mike at jerris.com>:
>
>> sofia is doing the right thing.. sendrecv is the default state if not
>> otherwise stated in the packet and is not necessary to send there. the
>> receiving device is broken if its not taking off hold there.
>>
>>
>> On Jul 24, 2017, at 5:18 AM, Dmitriy Borisov <borik.internet at gmail.com>
>> wrote:
>>
>> Hello!
>>
>> There are some issue (or not???) in mod_sofia module.
>>
>> You can reproduce this issue by holding call via 0.0.0.0 SDP:
>> v=0
>> o=OneXS 99765105 2 IN IP4 87.237.24.14
>> s=-
>> c=IN IP4 0.0.0.0
>> t=0 0
>> m=audio 49760 RTP/AVP 8 101
>> a=fmtp:101 0-15
>> a=rtpmap:101 telephone-event/8000
>> a=inactive
>>
>> and unholding via invite without SDP.
>>
>> In version 7a77e0b (branch v1.6) place of errorneus (as I think) code is
>> near:
>> 7641 if (switch_channel_var_true(channel,
>> "sip_unhold_nosdp")) {
>> 7642 switch_core_media_gen_local_sdp(session,
>> SDP_TYPE_RESPONSE, NULL, 0, "sendrecv",
>> 7643
>> zstr(tech_pvt->mparams.local_sdp_str) || !switch_channel_test_flag(channel,
>> CF_PROXY_MODE));
>> 7644 } else {
>> 7645 switch_core_media_gen_local_sdp(session,
>> SDP_TYPE_RESPONSE, NULL, 0, NULL,
>> 7646
>> zstr(tech_pvt->mparams.local_sdp_str) || !switch_channel_test_flag(channel,
>> CF_PROXY_MODE));
>> 7647 }
>> 7648
>> 7649 if (zstr(tech_pvt->mparams.local_sdp_str)) {
>> 7650
>> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_WARNING,
>> "Cannot find a SDP\n");
>> 7651 switch_channel_hangup(channel,
>> SWITCH_CAUSE_DESTINATION_OUT_OF_ORDER);
>> 7652 } else {
>> 7653 if (sofia_use_soa(tech_pvt)) {
>> 7654 *switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session),
>> SWITCH_LOG_WARNING, tech_pvt->mparams.local_sdp_str); // This is my add*
>> 7655 nua_respond(tech_pvt->nh, SIP_200_OK,
>> 7656
>> SIPTAG_CONTACT_STR(tech_pvt->reply_contact),
>> 7657
>> SOATAG_USER_SDP_STR(tech_pvt->mparams.local_sdp_str),
>> 7658 SOATAG_REUSE_REJECTED(1),
>> 7659 SOATAG_AUDIO_AUX("cn
>> telephone-event"),
>> 7660 TAG_IF(sofia_test_pflag(profile,
>> PFLAG_DISABLE_100REL), NUTAG_INCLUDE_EXTRA_SDP(1)), TAG_END());
>> 7661 } else {
>> 7662 nua_respond(tech_pvt->nh, SIP_200_OK,
>> 7663 NUTAG_MEDIA_ENABLE(0),
>> 7664
>> SIPTAG_CONTACT_STR(tech_pvt->reply_contact),
>> 7665
>> SIPTAG_CONTENT_TYPE_STR("application/sdp"),
>> SIPTAG_PAYLOAD_STR(tech_pvt->mparams.local_sdp_str), TAG_END());
>> 7666 }
>> 7667 }
>>
>>
>> In described situation code flow go throw line 7654 and post to
>> nua_respond correct SDP (with correct c=IN IP4 re.al.ip.addr and
>> a=sendrecv):
>> 2017-07-24 11:00:54.252668 [WARNING] sofia.c:7654 v=0
>> o=Sipean 1500875192 1500875195 IN IP4 x.x.x.x:
>> s=Sipean
>> c=IN IP4 x.x.x.x
>> t=0 0
>> m=audio 11642 RTP/AVP 8 101
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>> a=sendrecv
>>
>> sofia library sends SDP:
>> v=0
>> o=Sipean 1500875192 1500875195 IN IP4 x.x.x.x
>> s=Sipean
>> c=IN IP4 x.x.x.x
>> t=0 0
>> m=audio 11642 RTP/AVP 8 101
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>>
>> On SIP client side call in unholded state, but at FreeSWITCH side it is
>> still in HOLDED state
>>
>> I have experimented with nua_respond, but maximum result is a correct
>> call state on SIP client side (holded), but not unholding the line. Where
>> should I search for errors? Or this scenario can't work correct?
>>
>>
>> _________________________________________________________________________
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>>
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>
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